| 00:03.07 | *** join/#asterisk shinao1 (n=shinao1@41.219.238.74) |
| 00:04.22 | *** join/#asterisk Borealid (n=braujac@pool-71-163-70-123.washdc.east.verizon.net) |
| 00:05.10 | Borealid | hello all - is there anyone out there willing to help a reasonably experienced user set up TLS in Asterisk 1.6.0.9? |
| 00:09.41 | stope | I am willing to learn |
| 00:11.51 | Borealid | stope: well, all I've learned is that it doesn't come easily, if you were talking to me |
| 00:12.12 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 00:12.30 | Borealid | with a valid Godaddy SSL cert for the server, I've been unable to successfully get a SIPS connection going with any server or client |
| 00:12.40 | Borealid | 'bye all |
| 00:12.57 | *** part/#asterisk Borealid (n=braujac@pool-71-163-70-123.washdc.east.verizon.net) |
| 00:15.22 | VaGoNeTaS | [TK]D-Fender |
| 00:15.27 | VaGoNeTaS | i fix the Cel phone problem |
| 00:15.40 | VaGoNeTaS | it was boz of the fucking redfone shitty device |
| 00:15.50 | VaGoNeTaS | i had to add this line to the system.conf |
| 00:16.10 | KavanS | awww fuck |
| 00:16.15 | VaGoNeTaS | alaw=1-124 |
| 00:16.18 | VaGoNeTaS | that's it |
| 00:16.21 | VaGoNeTaS | no more chanunavail |
| 00:17.55 | Alborracho | im having trouble with dtmf un g729 can anyone help me? |
| 00:18.21 | [TK]D-Fender | telnettech: No, it should not be a "goto", that is unconditional and jsut jumps over other things. What would be the point? |
| 00:18.44 | telnettech | yeah i found that out the hard way |
| 00:19.06 | *** join/#asterisk nephfl (n=none@wsip-98-175-64-147.ga.at.cox.net) |
| 00:19.28 | telnettech | I still havent figured out what is wrong with the expression.....I am gonna continue but i went on and changed everything back and changedthe context of the extension in the voicemail database |
| 00:20.05 | nephfl | hello, im still trying to get my call files to work, i set zap channel as one end and an application as the other (voicemail) and chown it to asterisk, then mv it to the spool, and it doesnt usually work, it has once or twice |
| 00:20.11 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:21.22 | nephfl | can anyone tell me where i need to start looking? |
| 00:21.47 | telnettech | i think it should be as follows |
| 00:22.36 | telnettech | exten => 1198,n,GotoIf($(${CALLERIDNUM}="8888")?4:6) |
| 00:23.05 | telnettech | this is a very weak aspect of my dialplan building, i will admit it |
| 00:26.02 | carrar | shouldn't that be: |
| 00:26.02 | carrar | GotoIf($[${CALLERIDNUM} = "8888"]?4:6) |
| 00:26.06 | beek | GN all |
| 00:31.32 | VaGoNeTaS | no more chanunavail, no more noise |
| 00:36.45 | *** join/#asterisk trentcreek (n=kvirc@200.94.227.117) |
| 00:38.58 | carrar | hi |
| 00:40.33 | jaytee | hi |
| 00:40.48 | telnettech | thank you carrar..... |
| 00:41.38 | telnettech | I had that a couple times and was thinking that it was wrong cause i needed the () set and not the [] set |
| 00:41.54 | telnettech | even though it was biting me reading the book |
| 00:42.34 | telnettech | syntax, syntax, syntax |
| 00:42.45 | [TK]D-Fender | telnettech: $[] <------- EXPRESSION |
| 00:43.21 | [TK]D-Fender | telnettech: And you're using long dead variables, which is asking for trouble on upgrade |
| 00:43.51 | telnettech | I cant do anything....i am a lowly support tech who works on everything that is put in front of me |
| 00:44.17 | telnettech | I can recommend to upgrade as much as i want but I cant MAKE anyone do it |
| 00:45.06 | telnettech | It took me 6 long months just to convince my supervisor, the development mgr and the head developer to move to 1.4 and now i am trying to convince them to go to 1.6 |
| 00:45.09 | jaytee | yeah, it's almost harder to get some people to upgrade from 1.2 as it for some people to quit drinking |
| 00:45.36 | [TK]D-Fender | telnettech: Ask your dom if masochism may be right for you! |
| 00:45.54 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 00:46.16 | telnettech | there is alot of cloudy and murky wording in our sales contracts that no one seems to be able to tell if customer gets free upgrades of if they pay for them |
| 00:46.37 | telnettech | so they are not wanting to make too much until we get directives from HQ in Singapore |
| 00:47.19 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-07c3fc064d2d5ef1) |
| 00:47.19 | jaytee | the cloudiness should be rewritten to specify that upgrades are only free if the customer themselves installs it |
| 00:47.21 | telnettech | so i keep putting along |
| 00:47.34 | telnettech | i agree |
| 00:47.58 | telnettech | but at the same time there should be limit on how long we will support the old software |
| 00:48.13 | telnettech | before we "force" customer to get with the program |
| 00:49.15 | telnettech | but im an asshole!!!! |
| 00:50.07 | jaytee | yeah, and to get them to fix all the references to deprecated variables, functions and applications that need to be rewritten to run in 1.4 |
| 00:52.37 | *** join/#asterisk vuja (n=vvuja@77.105.29.83) |
| 00:52.39 | vuja | hello |
| 00:52.51 | vuja | I would like to appologize for my behaviour |
| 00:52.56 | vuja | last time I was here |
| 00:52.59 | vuja | nick was vvuja |
| 00:53.07 | vuja | and to ask for permission to stay on this channel |
| 00:53.11 | vuja | ? |
| 00:53.44 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 00:53.49 | carrar | [TK]D-Fender, yeah :) I should have corrected his variables too |
| 00:54.39 | carrar | GotoIf($[${CALLERID(number)} = "8888"]?4:6) |
| 00:55.36 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) |
| 00:56.03 | *** mode/#asterisk [-b vvuja!*@*] by Qwell |
| 00:56.09 | Qwell | vuja: ZERO tolerance. Get it? |
| 00:56.50 | carrar | heh |
| 00:58.20 | vuja | thank you Qwell |
| 00:58.39 | vvuja | I will try not to insult anyone if someone not insulting me |
| 00:59.48 | telnettech | thanks carrar |
| 00:59.57 | vvuja | and of course I would like to use this channel to learn new stuff about asterisk |
| 01:00.10 | telnettech | like i said: syntax, syntax, syntax |
| 01:00.48 | [TK]D-Fender | telnettech: You know you really should ahve found the answer on the wiki with a search every time I told you "asterisk expression". :) |
| 01:00.49 | carrar | telnettech, I would also avoid using priority numbers |
| 01:00.55 | carrar | Use labels |
| 01:01.30 | telnettech | i normally use the 'n' priority |
| 01:01.42 | carrar | yeah so use n(jumphere) |
| 01:01.49 | carrar | then change the 6 to something like jumphere |
| 01:01.53 | telnettech | oh |
| 01:02.16 | carrar | http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities |
| 01:02.18 | telnettech | i don t think that works in the version that is on this system |
| 01:02.29 | carrar | using version 1.0? |
| 01:02.32 | carrar | heh |
| 01:02.41 | telnettech | 1.2.4 |
| 01:02.47 | telnettech | close enough at this site |
| 01:02.54 | carrar | try it |
| 01:03.01 | telnettech | we have sites from that version all the way to 1.2.28 |
| 01:03.04 | carrar | The n Priority (Asterisk 1.2.x and later versions) |
| 01:03.19 | [TK]D-Fender | telnettech: You are also jumping to hard prioritys while using "n".... a shoot on sight offense :) |
| 01:03.44 | telnettech | lol |
| 01:04.04 | carrar | give the labels a try |
| 01:04.16 | *** part/#asterisk vvuja (n=vvuja@77.105.29.83) |
| 01:04.26 | *** join/#asterisk vvuja (n=vvuja@77.105.29.83) |
| 01:04.43 | vvuja | Qwell thank you |
| 01:04.47 | vvuja | see you again |
| 01:04.50 | *** part/#asterisk vvuja (n=vvuja@77.105.29.83) |
| 01:05.54 | telnettech | i like to do things the hard way....what else do you expect from someone learning as i go with no one really in house to guide me |
| 01:09.43 | telnettech | if it works then it works but if not then i try something else |
| 01:11.20 | telnettech | it may not pretty but it is effective |
| 01:14.06 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 01:17.57 | *** join/#asterisk Flyser_ (n=Flyser@unaffiliated/flyser) |
| 01:26.42 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 01:26.43 | *** join/#asterisk vvuja (n=vvuja@77.105.29.83) |
| 01:27.05 | vvuja | I asked this question before, but what do you think about openning for start free SIP server on the Internet? |
| 01:27.26 | vvuja | basicaly configured with softphones as clients |
| 01:27.30 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-bebbae2716cb6593) |
| 01:28.40 | [TK]D-Fender | vvuja: Many services already out there for that. |
| 01:29.06 | vvuja | I know |
| 01:29.15 | vvuja | trying to find something to do |
| 01:29.19 | [TK]D-Fender | vvuja: So why would anyone choose yours? |
| 01:29.23 | vvuja | of course |
| 01:29.25 | vvuja | there is Skype |
| 01:29.26 | vvuja | I know |
| 01:29.35 | [TK]D-Fender | vvuja: Skype is not SIP |
| 01:29.47 | vvuja | ok |
| 01:29.47 | [TK]D-Fender | vvuja: And aside from that though there ARE free SIP services out there |
| 01:30.01 | vvuja | ok but if I gather some people that I know |
| 01:30.07 | vvuja | and if with some donation they pay |
| 01:30.17 | vvuja | I can pay my dedicated server in USA for example |
| 01:30.27 | [TK]D-Fender | vvuja: Pay? You just said FREE. |
| 01:30.33 | vvuja | later, as donation |
| 01:30.35 | carrar | FreePay |
| 01:30.37 | vvuja | to keep server alive |
| 01:30.39 | carrar | Freepay.com |
| 01:30.41 | carrar | there you go |
| 01:30.43 | carrar | go you a name |
| 01:30.46 | [TK]D-Fender | vvuja: And since there are already free sevices out there why would anyone PAY for your service? |
| 01:30.54 | vvuja | ok I understand |
| 01:30.59 | vvuja | but it is just talking |
| 01:31.06 | carrar | oh already registered |
| 01:31.13 | vvuja | don't know how those free SIP server works |
| 01:31.27 | vvuja | quality and that stuff |
| 01:31.36 | carrar | so do it, make your own |
| 01:31.41 | carrar | make it VAULE ADD |
| 01:32.23 | vvuja | carrar: that was for me? |
| 01:32.26 | [TK]D-Fender | vvuja: What do you truely NEED to know? They're out there, go try them. |
| 01:32.31 | carrar | Will you be terminating and originating in Alaska soon vvuja |
| 01:32.33 | carrar | vvuja, yes |
| 01:32.37 | carrar | :) |
| 01:32.45 | [TK]D-Fender | vvuja: If you're bored and looking for something productive to do heres a thought : crochet |
| 01:32.57 | vvuja | carrar: You are interested? |
| 01:33.02 | vvuja | Fender: crochet? that is? |
| 01:33.08 | carrar | no |
| 01:33.18 | carrar | I'm already doing 'sip' |
| 01:33.33 | vvuja | carrar: ok it was just idea |
| 01:33.36 | carrar | yeah |
| 01:33.41 | vvuja | carrar: how to start bussines from home |
| 01:33.50 | carrar | don't let anyone stop you if you think you have some people to make money with |
| 01:33.59 | carrar | and you have the skills |
| 01:34.04 | vvuja | I know you will not support me, but simplest way is to pay with Visa Internet, and to receive payments on Visa Electron |
| 01:34.09 | vvuja | no company no anything |
| 01:34.13 | vvuja | just close circle of people |
| 01:34.20 | carrar | easy enough |
| 01:34.29 | vvuja | yes it is illegal |
| 01:34.34 | vvuja | but I live in Serbia, whatever |
| 01:34.38 | carrar | why is it illegal? |
| 01:34.47 | vvuja | that is what I was wondering |
| 01:34.51 | carrar | no |
| 01:34.59 | vvuja | how someone can check if I received 300 euros or dollars on my credit card |
| 01:35.02 | carrar | You will need to terminate 911 too |
| 01:35.02 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 01:35.11 | vvuja | carrar payed me because he loves me |
| 01:35.12 | carrar | or have a carrier who will |
| 01:35.14 | Micc | Anyone know of a good T38 ITSP that works with asterisk? |
| 01:35.15 | vvuja | or something like that |
| 01:35.21 | vvuja | carrar can we go on private |
| 01:35.23 | vvuja | ? |
| 01:35.32 | carrar | only if you talk HOT++ |
| 01:35.39 | vvuja | carrar: no no, for real |
| 01:35.48 | vvuja | carrar: I want to ask you something |
| 01:35.53 | carrar | ask away |
| 01:35.57 | vvuja | here? |
| 01:36.02 | carrar | or msg me |
| 01:36.05 | carrar | doesn't mater to me |
| 01:36.06 | vvuja | ok |
| 01:36.32 | carrar | Micc, MCI terminates T.38 |
| 01:37.12 | Micc | carrar, I thought MCI went out of business a while back. |
| 01:37.21 | carrar | ok call them verizon if you want |
| 01:37.34 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:38.50 | Micc | carrar, where can I find information about that? Is it on their website somewhere? |
| 01:39.17 | carrar | probably |
| 01:41.15 | carrar | Verizon wholesale voip |
| 01:41.19 | carrar | or something like that |
| 01:41.39 | carrar | probably can get T.38 cheaper someplace else however |
| 01:43.28 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 01:53.43 | Micc | So far, I'm running into t.38 companies that have shut down or stopped accepting signups. The first two sites I went to said they weren't accepting signups or you have to call. |
| 01:53.47 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:55.18 | saftsack | btw whats about asterisk and t.38= |
| 01:59.39 | *** join/#asterisk xpot (n=james@70.91.210.233) |
| 02:01.52 | thehar | i believe XO does our t.38 termination |
| 02:06.40 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 02:06.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:07.04 | *** part/#asterisk vvuja (n=vvuja@77.105.29.83) |
| 02:07.13 | Qwell | waves to Deeeeeeeeeeewayne |
| 02:07.45 | Deeewayne | Qwell, hey :-) |
| 02:08.34 | *** join/#asterisk jmoncayo (n=steven@190.152.163.166) |
| 02:15.07 | *** join/#asterisk saftsack (n=saftsack@p579242E9.dip.t-dialin.net) |
| 02:18.02 | Deeewayne | Qwell, you didn't happen to see a bunch of faxed "Ubuntu" pages on your printer today, did you ? |
| 02:18.27 | Qwell | mmm, nope |
| 02:18.55 | Deeewayne | hrm. :-/ |
| 02:24.06 | drmessano | carrar: You free sip server help me? |
| 02:25.04 | [TK]D-Fender | drmessano: Free for pay plan for business help I you will? |
| 02:26.38 | drmessano | Not sure why I would pay someone to access a SIP box with no termination or origination.. Just the joy of SIP |
| 02:26.43 | drmessano | HA.. thats it |
| 02:26.49 | drmessano | TheJoyOfSIP.com |
| 02:27.46 | stope | anyone setup aastra phones with SLA? I'm having caller id issues, the phone is only showing 'asterisk' as the cid info |
| 02:28.33 | stope | and when I put exten => s,n,SIPAddHeader(FROM: ${CALLERID(all)}) it won't ring but the blf lights light up when a call comes in |
| 02:29.26 | stope | if I put: exten => s,n,SIPAddHeader(cid: ${CALLERID(all)}) then the phone rings but shows 'asterisk' as the caller |
| 02:30.01 | stope | and it returns the following: Got SIP response 400 "Bad Request" back from 192.168.0.103 |
| 02:45.49 | *** join/#asterisk EricDGlobeTroote (n=GlobeTro@201.218.90.155) |
| 02:50.14 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.12) |
| 03:00.05 | *** join/#asterisk ctp_ (n=ctp@brsg-d9bee524.pool.mediaWays.net) |
| 03:05.23 | Micc | stope, are you setting the fromuser? |
| 03:05.33 | Micc | stope, in sip.conf. |
| 03:06.19 | Micc | stope, don't use fromuser, it will overwrite the callerid. |
| 03:06.39 | stope | from user is blank in sip.conf |
| 03:06.51 | stope | fromdomain also blank |
| 03:07.23 | stope | it works ok w/the snom phone, but the aastra isn't playing nice |
| 03:08.18 | *** join/#asterisk jmacz (n=jmacz@cable201-233-192-36.epm.net.co) |
| 03:09.14 | *** join/#asterisk uTx (n=unix@96.23.1.195) |
| 03:10.06 | Micc | stope, I've never used SLA with asterisk since [TK]D-Fender says its a hack. |
| 03:10.38 | Micc | stope, but the aastra phones I like a lot. Especially the new one 6731i. |
| 03:10.45 | Micc | Make sure you upgrade to the latest firmware though. |
| 03:10.50 | Micc | The first release was pretty buggy. |
| 03:11.08 | stope | hmm... great... looking for the update now... |
| 03:15.15 | stope | updating the little bugger now.... we'll see if it helps :) |
| 03:16.53 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 03:19.38 | stope | Micc firmware updated... same shit |
| 03:19.55 | stope | all it wants to show is 'asterisk' |
| 03:28.55 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 03:29.17 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 03:30.08 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 03:30.11 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 03:38.49 | Micc | stope, when you make calls between extensions does it show asterisk as the caller? |
| 03:38.57 | Micc | stope, or is it only when your using sla? |
| 03:41.40 | *** join/#asterisk blkry (n=blkry@97.95.233.232) |
| 03:46.59 | *** join/#asterisk chendy (n=chatzill@58.251.102.216) |
| 04:06.52 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 04:20.50 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 04:39.48 | *** join/#asterisk juanIMP (n=Juancho@201.245.248.194) |
| 04:41.52 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 04:43.41 | joako | I have two things I want to implement, can anyone offer advice? 1) Roaming logins, i.e. employee is assigned an extension number and they can login to any phone and get their calls there 2) Pop-up on inbound call a URL that contains the caller ID number |
| 04:47.03 | [TK]D-Fender | joako: 1st is easy, its all dialplan. manke an exten to auth the login. Once accepted put the device name that called in a DB entry associating with the "exteension" they authed as. Have your dialplan to all "roaming" extensions check what device they are authed to. |
| 04:47.31 | [TK]D-Fender | joako: #2 requires some rel program running on the client |
| 04:47.34 | [TK]D-Fender | real* |
| 04:48.21 | joako | [TK]D-Fender: I understand that. Also, I believe HUD or HUDlite can support that, but is there anything that will run on Linux to do that? |
| 04:48.39 | [TK]D-Fender | joako: Nothing "ready to use" that I know. |
| 04:48.57 | joako | Anything sort of ready to use? |
| 04:49.13 | [TK]D-Fender | ... |
| 04:49.46 | [TK]D-Fender | joako: Don't go "minor semantic" on me... |
| 04:52.43 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 04:59.44 | *** join/#asterisk cspiess (n=Connor@c-76-113-167-208.hsd1.mn.comcast.net) |
| 05:12.51 | *** join/#asterisk s0lid (n=s0lid@122.53.104.218) |
| 05:15.52 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-183-141.telkomadsl.co.za) |
| 05:18.38 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 05:18.49 | b14ck | hi all |
| 05:19.01 | b14ck | do any of you knwo if it is possible to modify a dial plan LIVE, without modifying the file and reloading? |
| 05:19.08 | b14ck | and if so--how? heh |
| 05:20.12 | juanIMP | you coult use Asterisk Real-Time b14ck "ARA" |
| 05:20.23 | b14ck | ok thanks, i'll google that |
| 05:20.34 | juanIMP | ok b14ck |
| 05:23.22 | *** join/#asterisk n00m (n=n00m@c-67-167-200-89.hsd1.il.comcast.net) |
| 05:24.17 | juanIMP | something like this b14ck http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip for sip peers, but you also need a front end to configure it.....if you dont want query a lot with the MySql prompt |
| 05:24.32 | joako | [TK]D-Fender: After some research FOP supports what I need to open the web-based CRM. There is also ASTassistant which probably could run with WINE |
| 05:25.04 | juanIMP | or create a MySql shell :D b14ck |
| 05:31.43 | *** part/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 05:40.38 | *** join/#asterisk mbuf (n=user@61.16.248.242) |
| 05:40.54 | blkry | I have a remote polycom through a VPN that shows in FreePBX that is online but, shows unreachable in SIP show peers. I have searched and searched. |
| 05:41.41 | blkry | I have read forum after forum and no seem to have a good answer |
| 05:43.11 | blkry | none seem to have a good answer. |
| 05:49.30 | *** join/#asterisk obnauticus (i=obnautic@about/windows/regular/obnauticus) |
| 05:49.49 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 05:54.24 | mbuf | does anyone recommend a specific documentation for testing asterisk with VoIP? i have been STW for some time now; |
| 05:54.33 | mbuf | *setting it up and testing |
| 05:54.35 | b14ck | sure |
| 05:54.50 | *** join/#asterisk Titan8990 (n=Titan899@unaffiliated/titan8990) |
| 05:55.07 | Qwell | ~book |
| 05:55.07 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 05:55.13 | Qwell | mbuf: ^^ |
| 05:55.32 | b14ck | ya i agree with Qwell, i was just googling the link for it now :) |
| 05:55.36 | *** part/#asterisk Titan8990 (n=Titan899@unaffiliated/titan8990) |
| 05:55.37 | b14ck | It's an awesome book ^^ |
| 05:56.22 | mbuf | Qwell: blkry thanks |
| 06:04.28 | *** part/#asterisk mbuf (n=user@61.16.248.242) |
| 06:10.53 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:21.06 | blkry | i read about where a remote phone works once and it never works again. I think I have experienced that tonight. Zoiper said "waiting for answer" a couple times. then unreachable and no audio |
| 06:21.29 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 06:23.11 | blkry | any ideas. its a Juniper VPN. switch to Liksys? |
| 06:23.54 | blkry | what is cheapish and can do a dynamic VPN? |
| 06:28.24 | *** join/#asterisk xrmx__ (n=rm@host119-200-dynamic.180-80-r.retail.telecomitalia.it) |
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| 07:20.36 | trafim | hi. i bought ABE license, and now a bit confused: what's the difference between A, B and C releases? i couldn't find the answer at digium site |
| 07:26.09 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
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| 07:50.50 | *** part/#asterisk trafim (n=reallyma@95.37.179.113) |
| 07:56.23 | *** join/#asterisk desdesdesdes (n=f@196.211.34.2) |
| 07:56.27 | *** join/#asterisk frantic667 (n=toffifee@dsbg-4db5f7d4.pool.einsundeins.de) |
| 07:56.39 | frantic667 | good morning |
| 07:56.57 | desdesdesdes | i can not get anyone to link via sip to my asterisk server, i have have an static address, and i have opened port 5060 too and from my server. |
| 07:57.07 | desdesdesdes | frantic667 MORNING |
| 07:57.53 | frantic667 | desdesdesdes are the ports really forwarded? did you try a portscan? |
| 07:58.01 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 07:59.15 | desdesdesdes | i have 2 asterisk servers linking with iax2 port 4569 works like a charm, but for some reason unknown to me sip port 5060 does not wanna work |
| 07:59.21 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:59.56 | frantic667 | I have a problem with the german provider 1und1, is here someone who has a working config for me? the problem seems to be that the calls are from multiple sip gateways... |
| 08:00.50 | frantic667 | desdesdesdes is the bindport variable set to 5060? are you able to reach the port from localhost? |
| 08:01.29 | frantic667 | may you paste your sip.conf to a pastebin? |
| 08:09.20 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:16.52 | desdesdesdes | yes my bind port is set 5060 |
| 08:17.14 | desdesdesdes | do i need to add any additional ports for remote sip connections? |
| 08:18.25 | frantic667 | you need to forwart the port range defined in the rtp.conf, sip only initiates the calls, the voice is transmittet via rtp protocol |
| 08:19.42 | desdesdesdes | doesrtp refer to ports on local server |
| 08:20.43 | desdesdesdes | so you are saying i need to port forward ports 10000 to 20000 to asterisk server? |
| 08:20.49 | frantic667 | yes, if you don't forward the rtp ports you won't hear anything in the best case. |
| 08:21.43 | desdesdesdes | can that affect not linking at all, error on registering from softphone from remote client to local asterisk server |
| 08:22.09 | frantic667 | yes, you may change the range, calculate about 2 or 4 ports per client. so if you only have one sip phone, i think it would be enough to define a range from 10000 to 10010 |
| 08:22.25 | desdesdesdes | does iax also use these ports |
| 08:22.45 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 08:22.47 | frantic667 | i have no experience in iax, but I don't think so |
| 08:22.54 | desdesdesdes | thx let me give this a go |
| 08:23.03 | frantic667 | good luck ;-) |
| 08:23.32 | desdesdesdes | ta |
| 08:24.00 | desdesdesdes | sorry one more question is this tcp or udp trafic? |
| 08:27.01 | pif | tzafrir_laptop: hi, could you give me your debian repository's address? |
| 08:28.17 | frantic667 | usually it's udp |
| 08:32.00 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 08:32.14 | doolittlework | frantic667: no luck |
| 08:32.45 | doolittlework | anyone wanna try and link with sip connection to my server for testing please |
| 08:33.11 | frantic667 | i can try, of course |
| 08:33.57 | doolittlework | thx ::extension 2000 password 1234 ip: 196.211.34.2 |
| 08:35.22 | frantic667 | [May 19 10:35:02] WARNING[14597]: chan_sip.c:15272 handle_response_register: Got 200 OK on REGISTER, but there isn't a registry entry for '2000' (we probably already got the OK) |
| 08:35.35 | doolittlework | let me see |
| 08:35.41 | frantic667 | status is "registered" |
| 08:35.48 | frantic667 | tha was my asterisk, by the way |
| 08:36.45 | doolittlework | k it works |
| 08:36.53 | doolittlework | thnx dude |
| 08:36.58 | frantic667 | no problem ;-) |
| 08:37.04 | doolittlework | can i try and phone u? |
| 08:37.07 | tzafrir_laptop | pif, http://updates.xorcom.com/pkg-voip/ |
| 08:37.44 | frantic667 | sorry, thats not possible, i'm at work and here is a connection about 10kbps... |
| 08:38.01 | doolittlework | lol np thx for your help |
| 08:38.05 | frantic667 | that's why i registered with my asterisk server^^ |
| 08:38.14 | frantic667 | รยฎ'm gla i could help you ;) |
| 08:40.29 | *** join/#asterisk plq (n=plq@78.189.30.218) |
| 08:42.34 | *** join/#asterisk plq (n=plq@78.189.30.218) |
| 08:43.59 | pif | tzafrir_laptop: thx |
| 08:47.46 | doolittlework | works like a charm thx frantic667 |
| 08:48.27 | frantic667 | doolittlewor: i'm happy to hear that :-) |
| 08:49.18 | frantic667 | doolittlework: i'm happy to hear that :-) (sorry for the missing "k) |
| 08:57.54 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6048ef243bbd2612) |
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| 09:11.52 | *** join/#asterisk jchillerup (n=aaa@hald.gbar.dtu.dk) |
| 09:12.36 | jchillerup | Is there a good beginner's guide for asterisk. Initially I just want it to listen to a SIP connection and play a song if somebody calls. |
| 09:16.55 | Chainsaw | jchillerup: For playing the song you could use the music-on-hold feature. |
| 09:17.37 | jchillerup | Yeah. But I haven't found a guide on how to make it connect to a SIP server |
| 09:17.50 | Chainsaw | jchillerup: Generally you would be the SIP server yourself. You just have a client connecting. |
| 09:18.17 | Chainsaw | jchillerup: People then find you by you having an SRV record in the DNS zone for say... jchillerup.net |
| 09:18.42 | jchillerup | Oh, but is it possible for asterisk to connect to a SIP server and claim a VoIP telephone numbre from that, as if it was just a softphone |
| 09:19.01 | jchillerup | number* |
| 09:19.10 | Chainsaw | Asterisk can pretty much do anything you can think of, as long as you manage to configure it. |
| 09:20.08 | *** join/#asterisk shinao1 (n=shinao1@41.219.248.87) |
| 09:20.20 | jchillerup | ... which I can't :P |
| 09:20.35 | jchillerup | Any advise on a go-to book/guide/article about setting up a system like that |
| 09:20.40 | jchillerup | i mean, it's pretty simple |
| 09:21.02 | jchillerup | advice* |
| 09:31.21 | *** join/#asterisk asim- (n=asim@mx1.beatthatquote.com) |
| 09:32.24 | asim- | Hey, can anyone help me out. I'm trying to troubleshoot a problem with our asterisk server. People are complaining that there is distortion/static on the line with all calls. I've tried replacing handsets, switches, etc. The cpu is idle on asterisk server. I cant figure it out. |
| 09:32.27 | asim- | Any suggestions? |
| 09:54.30 | *** join/#asterisk saftsack (n=saftsack@p579242E9.dip.t-dialin.net) |
| 09:59.57 | *** join/#asterisk hwt (i=hwt@curb.thorkildssen.com) |
| 10:00.08 | hwt | is ${MACRO_EXTEN} deprecated in 1.4? |
| 10:00.13 | hwt | what's the variable called now? |
| 10:01.14 | plundra | Hmm, is there an easy application/cmd, like Echo but with a delay? (Record n seconds, then play it back) |
| 10:01.31 | *** join/#asterisk uluatu (n=uluatu@189.32.44.21) |
| 10:14.20 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 10:14.25 | plundra | God damn Record, I'm going nuts. Whatever I do, it returns directly :-( |
| 10:15.34 | plundra | slaps himself. |
| 10:17.22 | plundra | Record() Plays a beep it self :-) So I got beep, beep and I assumed it was the beep after my Record() that got played . |
| 10:18.58 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
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| 10:40.41 | *** join/#asterisk doolittlework (n=f@196.211.34.2) |
| 10:41.38 | doolittlework | hi there i have a remote sip conection to my asterisk server |
| 10:41.49 | doolittlework | my * server is behind a firewall |
| 10:42.21 | doolittlework | i can see the phone want to register |
| 10:42.47 | doolittlework | but it says unreachable |
| 10:43.41 | doolittlework | i have opened port 5060 and 10000 to 20000 is there any other ports i should know about to get the remote sooftphone to work of my * server |
| 10:56.41 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
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| 11:19.51 | *** join/#asterisk aphexer (n=aphexer@jig.ulyssis.student.kuleuven.be) |
| 11:21.22 | aphexer | I got a registered phone number so that asterisk is reachable through eg 1234567. Now I want to make it that when someone calls 1234567# with # any digit, that the call is automatically connected to some extension. In other words, how can I find out the number which was originally called from within asterisk? |
| 11:21.41 | *** join/#asterisk stope (n=nobody@chelmsford-cable-69-60-242-213.unitz.ca) |
| 11:26.39 | aphexer | When I call 1234567# myself, I get connected to asterisk, so the regular phone company does forward it and doesn't care there's an extra digit. But I'm not sure that it's telling me the real number which was called... |
| 11:46.14 | freh | aphexer: define an extension in your incoming context |
| 11:47.06 | freh | exten => 12345674,1,Dial(SIP/1000) |
| 11:49.36 | *** join/#asterisk proxium (n=proxium@196.203.51.238) |
| 12:00.05 | stope | anyone know why a linksys phone would return the following: Got SIP response 486 "Busy Here" back from ... if its not in DND mode..? |
| 12:07.18 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:13.15 | *** join/#asterisk phurl (n=mdupont@mwhaeh.static.otenet.gr) |
| 12:20.43 | aphexer | freh: but all incoming calls come in on the same context and extension... |
| 12:21.46 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
| 12:22.22 | aphexer | I use the register command in sip.conf to get all incoming calls to go to some extension, so they all end up on the same place... |
| 12:24.10 | phurl | what is the best asterisk predictive dialer? |
| 12:27.15 | doolittlework | hi there i have a proble with a remote sip softphone connecting trhough the internet to my asterisk server, i can hear the phone ring but if they answer on the other side it just keeps ringing on my side |
| 12:27.48 | doolittlework | i set qaulify = no |
| 12:33.57 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-bcef73ee98d24271) |
| 12:34.34 | *** join/#asterisk w9sh (n=sph@68.209.117.205) |
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| 12:39.22 | angryuser | hey guys, i have a person who have bought 98 g729 license from digium, the thing is that they changed server recently and they are unable to reinstall it twice coz the first activation is attached to mac(?) address, is there any procedure for receting things ? |
| 12:39.44 | angryuser | do i need to call digium ? |
| 12:40.29 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:40.58 | doolittlework | download open g729 codes |
| 12:41.43 | doolittlework | you can contact them and i am sure the can help u |
| 12:41.47 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 12:41.51 | ariel_ | angryuser: the correct way is for you to call digium and they will help through the switch |
| 12:42.34 | doolittlework | i had the same prob before, one of my servers crashed and they helped me no problem |
| 12:42.41 | angryuser | ariel_, tryed to call them their ivr does not list g729 as product , and it tell me that i dont have a support plan xD |
| 12:43.08 | doolittlework | just go to normal support they will transfer u to right department |
| 12:43.19 | [TK]D-Fender | angryuser: thats what "press 0 to speak with an operator" is for :) |
| 12:43.38 | angryuser | [TK]D-Fender, ah, forgot the magic button! |
| 12:43.39 | [TK]D-Fender | angryuser: just get ANY human and you'll be fine. |
| 12:44.25 | *** join/#asterisk ctaloi (n=Adium@nat-66-218-1-29.usadatanet.com) |
| 12:44.26 | [TK]D-Fender | angryuser: IVR's are good for 2 levels of menu tops, with guaranteed ways of getting an operator and direct exsions to employees |
| 12:44.36 | doolittlework | why can i not get voice to work on remote sip,, the phone rings on the othere side but i dont get any voice accross, the other problem is that if i call the remote number it just rings, if they answer on the other side we can not conect? |
| 12:45.04 | ariel_ | doolittlework: nat issues |
| 12:45.17 | [TK]D-Fender | ~sipnat |
| 12:45.18 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:45.20 | [TK]D-Fender | ^^^^^ |
| 12:45.22 | *** join/#asterisk hi365 (n=hi365@94.159.178.51) |
| 12:45.29 | angryuser | it is not too early for US now ? hm |
| 12:45.57 | ariel_ | early ? |
| 12:46.07 | *** part/#asterisk ctaloi (n=Adium@nat-66-218-1-29.usadatanet.com) |
| 12:46.14 | *** join/#asterisk ctaloi (n=Adium@nat-66-218-1-29.usadatanet.com) |
| 12:46.38 | angryuser | ariel_, yes i dont know in what time zone they are |
| 12:46.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 12:46.59 | angryuser | the operator is out anyway |
| 12:47.20 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
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| 12:49.27 | *** part/#asterisk ctaloi (n=Adium@nat-66-218-1-29.usadatanet.com) |
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| 12:57.40 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 12:58.41 | [TK]D-Fender | Plattipus: Spelling is a little off... better fix it before jaytee slaps you in a cage as a new species! |
| 12:58.56 | [TK]D-Fender | Ah the noble platypus .... living proof that God DOES have a sense of humour... |
| 12:59.11 | jaytee | and is handy with spare parts |
| 12:59.12 | Plattipus | correct spellings were taken.....!!! |
| 12:59.41 | Plattipus | sorry |
| 13:00.38 | Plattipus | lol, yes, indeed he does |
| 13:03.45 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:05.09 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 13:05.50 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
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| 13:07.32 | Katty | it's going to be a weird day. coming out of the gas station after gettign my usual caffeine, some old guy from the next door car wash was standing next to my car. |
| 13:07.40 | Katty | he said, just so you know, there's no dumping. |
| 13:07.47 | Katty | i was like...whattt? are you talknig about? |
| 13:08.06 | Katty | then he turned around and walked back to the car wash |
| 13:08.16 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 13:08.26 | Katty | i'm still not sure what all that was about. |
| 13:12.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:16.33 | jasonwoot | do not try and dump your garbage; that's impossible |
| 13:16.44 | jasonwoot | instead only try and realize the truth: there is no dumping |
| 13:16.59 | jasonwoot | then you will see that it is not the garbage that gets dumped, it is yourself |
| 13:17.09 | jasonwoot | that's probably what he was trying to say |
| 13:18.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:18.54 | Katty | i think you missed the point that i'd just gone into the gas station to get a soda. |
| 13:19.01 | *** join/#asterisk thuglife (i=robert@blade2k.humppa.hu) |
| 13:19.12 | Katty | i think something was /wrong/ with him |
| 13:19.28 | thuglife | hello. i've a phone configured in sip.conf as nat=yes but in the sip show peers list it's say N in the NAT column |
| 13:19.42 | jasonwoot | what's really going to bake your noodle later is, would you have still broken the vase if he hadn't told you about the garbage |
| 13:19.44 | thuglife | and it does not work either because it tries to connect to the phone's internal ip address |
| 13:20.03 | Katty | it's going to be a crazy day. |
| 13:20.12 | Katty | admits self to the asylum. |
| 13:21.22 | [TK]D-Fender | thuglife: pastebin the SIP debug and your config masking only passwords. |
| 13:21.24 | [TK]D-Fender | ~pb |
| 13:21.25 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 13:21.28 | [TK]D-Fender | ^^^^^^ |
| 13:21.59 | [TK]D-Fender | bludgeons jasonwoot with a spoon |
| 13:22.17 | *** join/#asterisk ZefK (n=ZefK@wsc-fo.b.astral.ro) |
| 13:22.54 | jasonwoot | Katty, what you should take away from the experience is, why the hell are you drinking soda for breakfast? |
| 13:23.13 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 13:23.17 | Katty | ^_- |
| 13:23.21 | Katty | i always drink soda in the morning. |
| 13:23.48 | Katty | it's no mystery to me |
| 13:26.49 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.138.206) |
| 13:26.53 | thuglife | [TK]D-Fender: http://pastebin.ca/1427436 |
| 13:27.55 | [TK]D-Fender | thuNo internal IP's there... |
| 13:28.04 | [TK]D-Fender | thuglife: Though the rest of your NAT setup is incomplete |
| 13:28.10 | [TK]D-Fender | ~sipnat |
| 13:28.10 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:28.16 | [TK]D-Fender | thuglife: ^^^ |
| 13:29.35 | thuglife | [TK]D-Fender: asterisk is not behind nat |
| 13:29.39 | thuglife | the phones are |
| 13:29.49 | thuglife | and sometimes asterisk tries to connect to an internal address |
| 13:30.47 | [TK]D-Fender | thuglife: Not in that debug it doesn't, and even if * itself has a public IP it is still not set right |
| 13:32.03 | [TK]D-Fender | thuglife: [efone] should be "nat=no" |
| 13:32.04 | thuglife | 213.163.8.130 is the right address and that's what i see there |
| 13:32.21 | [TK]D-Fender | thuglife: So everything you've shown is working fine so far... |
| 13:32.31 | [TK]D-Fender | thuglife: Except that your ISP entry should be "nat=no" |
| 13:32.41 | ZefK | Hi, I saw an old information (http://lists.digium.com/pipermail/asterisk-dev/2003-September/001751.html) that there is a patch for libpri to display the UUI field. Is it possible to access UUI info from dialplan? Thanks. |
| 13:33.03 | thuglife | [TK]D-Fender: ok but then why traffic stops flowing after i asnwer the call? |
| 13:33.10 | thuglife | no audio et all |
| 13:33.23 | [TK]D-Fender | thuglife: Show a complete call, and like I said, your ITSP entry is bad |
| 13:33.31 | *** join/#asterisk joesuffceren (n=chatzill@ip68-104-166-24.ph.ph.cox.net) |
| 13:33.46 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 13:33.53 | thuglife | [TK]D-Fender: i fixed that |
| 13:33.56 | thuglife | testing it soon |
| 13:34.03 | [TK]D-Fender | thuglife: check your firewall on the server |
| 13:37.02 | joesuffceren | I have a 21 channel PRI on a digium TE210. 3 times in the past 2 weeks my incoming calls have started getting busy signals even though there are only 4-6 PRI channels in use. My telco tells me that my pbx is telling his equipment "busy here" when the calls come in. When I place an incoming call and watch the CLI (set on verbose 9) I don't get any indication that a call is coming in. |
| 13:37.22 | joesuffceren | Outgoing calls work fine |
| 13:37.34 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 13:37.47 | [TK]D-Fender | joesuffceren: enable PRI debug and watch. Barring that, pastebin your configs and status dumps |
| 13:39.31 | timeshell_atwork | [TK]D-Fender : Just as an FYI to you, the loop issue turned out to be an Asterisk bug. |
| 13:39.55 | joesuffceren | [TK]D-Fender: here's the pri debug for one incoming call: http://pastebin.ca/1427446 |
| 13:40.10 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 13:40.23 | timeshell_atwork | Which I finally was able to point a finger to directly on Friday evening. |
| 13:40.37 | mvanbaak | hides for lesouvage |
| 13:41.45 | lesouvage | mvanbaak: why? |
| 13:45.27 | doolittlework | dumb question time: if the phone can get the assigned nmber on his softphone, does that mean he is registered on the local asterisk server? |
| 13:48.04 | *** join/#asterisk propellerhead (n=yogurt2u@host44.190-136-118.telecom.net.ar) |
| 13:49.00 | therealcircut | morning all |
| 13:49.01 | [TK]D-Fender | doolittlework: what does "get the assigned nmber on his softphone" mean? |
| 13:49.48 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 13:51.00 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 13:51.06 | doolittlework | [TK]D-Fender: i created extension 212121 on my asterisk server the phone is a softphone on the remote side it get the number on the softphone screen but on my asterisk server it still comes up as Unmonitored |
| 13:51.34 | [TK]D-Fender | doolittlework: "unmonitored" means you did not enable Qualify on that peer |
| 13:51.49 | [TK]D-Fender | doolittlework: And * does not check to see that it is still alive or send NAT keep-alives, etc |
| 13:52.08 | joesuffceren | [TK]D-Fender: from the PRI debug it looks like my telco is requesting channel 44 which my asterisk doesn't have, and my pbx is saying channel unavailable. does that sound like a telco issue? |
| 13:52.24 | doolittlework | i created port forwarding on the remote router for incoming 5060 udp trafic to go to the computer ip and port where the softphone is installed on. |
| 13:52.35 | [TK]D-Fender | joesuffceren: I still don't have your confifgs.... |
| 13:52.51 | [TK]D-Fender | doolittlework: You do not need to port forward for remote phones. |
| 13:53.02 | *** join/#asterisk fallenwrk (n=fallen@93-80-181-168.broadband.corbina.ru) |
| 13:53.06 | fallenwrk | hello |
| 13:53.08 | joesuffceren | [TK]D-Fender: sorry. I thought you wanted PRI debug or configs. one sec |
| 13:53.24 | [TK]D-Fender | doolittlework: Just enable quality and disable reinvites |
| 13:54.08 | doolittlework | quality of qualify? |
| 13:54.12 | doolittlework | or |
| 13:55.04 | *** join/#asterisk BobPierce (n=BobPierc@216.36.132.162) |
| 13:55.35 | fallenwrk | i ve asterisk 1.6.0.9. Trying to call with agi script for sip. But the call is hanguped, and there is error in log: [May 19 13:32:55] ERROR[1509] utils.c: write() returned error: Broken pipe |
| 13:55.42 | fallenwrk | what does it mean? |
| 13:56.12 | *** join/#asterisk ssokol (n=ssokol@ip70-189-74-33.ok.ok.cox.net) |
| 13:56.25 | [TK]D-Fender | doolittlework: "qualify=yes" |
| 13:56.29 | [TK]D-Fender | typo'd |
| 13:56.37 | mvanbaak | lesouvage: just because I was going out for a smoke when you came in :) |
| 13:56.43 | doolittlework | [TK]D-Fender now it says 12121/212121 196.210.129.194 D A 2051 UNREACHABLE |
| 13:57.06 | doolittlework | why is picking up port 2051? |
| 13:57.15 | joesuffceren | [TK]D-Fender: zaptel.conf: http://pastebin.ca/1427455 |
| 13:58.06 | doolittlework | joesuffceren link not working, try pastebin.com |
| 13:59.16 | joesuffceren | zapata.conf: http://pastebin.com/m6d9785c4 |
| 13:59.26 | *** join/#asterisk ssokol (n=ssokol@ip70-189-74-33.ok.ok.cox.net) |
| 13:59.42 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 14:01.18 | doolittlework | joesuffceren paste your zaptel.conf |
| 14:02.48 | joesuffceren | doolittlework: http://pastebin.com/m7fe3c1bd |
| 14:02.57 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 14:03.31 | doolittlework | do yo uhave 2 pri joesuffceren? |
| 14:04.08 | joesuffceren | the card has the capability for 2. I use only 1. |
| 14:05.12 | doolittlework | what do u get if you tipe : pri shoe spans from asterisk CLI? joesuffceren |
| 14:05.13 | joesuffceren | doolittlework: PRI span 1/0: Provisioned, Up, Active |
| 14:06.06 | doolittlework | so you are having problems with incoming calls? |
| 14:06.14 | fallenwrk | i ve asterisk 1.6.0.9. Trying to call with agi script for sip. But the call is hanguped, and there is error in log: [May 19 13:32:55] ERROR[1509] utils.c: write() returned error: Broken pipe |
| 14:06.53 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 14:06.59 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) |
| 14:08.19 | doolittlework | joesuffceren paste cli for incoming call please |
| 14:08.55 | Katty | file: eleanor rigby! |
| 14:09.06 | file | Katty: ^_^ |
| 14:09.16 | Katty | file: he's from missouri, apparently. |
| 14:09.19 | Katty | file: not too far from here |
| 14:09.55 | [TK]D-Fender | joesuffceren: < Ext: 1 Channel: 21 ] <- might be an issue if they are 0-based and that actually means #22 |
| 14:09.59 | jasonwoot | any isymphony users know of a way to report on how many users are running the software? |
| 14:10.07 | [TK]D-Fender | joesuffceren: Though you see NOTHING else with max debug & verbose? |
| 14:10.16 | jaytee | mornin Katty *hugs* |
| 14:10.23 | Katty | allo jaytee (= |
| 14:10.25 | Katty | hugs on jaytee |
| 14:10.36 | joesuffceren | doolittlework: I get nothing onthe CLI for an incoming call even when verbose is set to 9. I posted the PRI debug to pastebin.com for you: http://pastebin.com/md91454a |
| 14:11.39 | cjk | hi, i have several installations and on one installations phones are getting randomly UNREACHABLE for exactly 10 seconds. any ideas on how to find the cause? |
| 14:11.43 | joesuffceren | [TK]D-Fender: I issued pri debug span 1. Is that not "max" debug? |
| 14:12.00 | [TK]D-Fender | joesuffceren: for PRI side, I'm talking the rest of * |
| 14:12.34 | joesuffceren | I was on "core set verbose 9" I got nothing at all until I enabled pri debug |
| 14:12.36 | [TK]D-Fender | cjk: Sounds like networking issues, most common of which is improper NAT setup |
| 14:13.05 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:13.29 | doolittlework | joesuffceren: paste custom-cid-spoofing context in extensions.conf |
| 14:13.33 | cjk | [TK]D-Fender, no NAT in the row |
| 14:13.41 | cjk | local network |
| 14:13.57 | [TK]D-Fender | cjk: Local firewalls then |
| 14:15.03 | cjk | [TK]D-Fender, no firewalls, just a switch a phone and a pbx, well several switches and sveral phones |
| 14:15.15 | *** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do) |
| 14:15.17 | [TK]D-Fender | cjk: I'm talking on the * itself |
| 14:16.15 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:16.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:16.35 | cjk | [TK]D-Fender, no iptables running, so im quite lost, the setup is not complex |
| 14:16.57 | joesuffceren | doolittlework: that's just a little dialplan to change my cell phone number so my users don't get my cell phone. I munged the phone numbers, but here it is if you care: http://pastebin.com/mc72e218 |
| 14:17.23 | [TK]D-Fender | cjk: Pastebin CLI output, SIP debug, configs, etc.... |
| 14:17.58 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 14:21.14 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:21.29 | cjk | [TK]D-Fender, hmm difficult to find the moment when it happens |
| 14:21.55 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 14:22.19 | *** join/#asterisk tegg (n=aspod@mnhm-5f75df58.pool.einsundeins.de) |
| 14:25.03 | *** join/#asterisk smps (n=smps@193.170.53.51) |
| 14:25.33 | tegg | hi |
| 14:25.58 | tegg | i'm new in asterisk |
| 14:26.20 | tegg | and have a lot of "Warning[27335]" |
| 14:26.26 | tegg | when i start asterisk |
| 14:26.38 | tegg | 1.4.22 Debian |
| 14:26.59 | tegg | and i found nothing with google |
| 14:27.19 | fallenwrk | ignore warnings |
| 14:27.36 | tegg | ok |
| 14:27.52 | tegg | but asterisk ends |
| 14:27.58 | tegg | and didn't work |
| 14:28.39 | tegg | last output is |
| 14:28.43 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:28.43 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:28.49 | tegg | asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_esel.so: undefined symbol: ast_extension_state_callerid_add |
| 14:29.10 | fallenwrk | config sintax error |
| 14:29.30 | fallenwrk | i think |
| 14:29.53 | file | res_esel is not part of the Asterisk standard tree |
| 14:30.11 | tegg | i don't know |
| 14:30.11 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 14:30.24 | tegg | i use the apt-get version from debian |
| 14:31.04 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 14:31.08 | ck_28 | hi ppl |
| 14:31.24 | tegg | an tried to connect a ISDN Phone with asterisk |
| 14:31.42 | ck_28 | i have a problem with open Source G.729 (g729) with Xion Cpu |
| 14:31.44 | tegg | with this tutorial http://bach-online.de/blog/?p=50 |
| 14:32.23 | ck_28 | what type of codec i should install like codec_g729-ast16-gcc4-glibc-pentium4.so |
| 14:32.34 | Chainsaw | ck_28: It depends on what kind of Xeon it is. |
| 14:32.45 | Chainsaw | ck_28: Some Xeons are P3-based, some P4, some Core2. |
| 14:33.04 | ck_28 | Chainsaw model name : Intel(R) Xeon(TM) CPU 3.40GHz |
| 14:33.29 | Chainsaw | Based on the speed rating, P4-based. Are you running a 64-bit or 32-bit kernel please? |
| 14:33.36 | Chainsaw | uname -a output should tell you. |
| 14:33.39 | ck_28 | Chainsaw 32 bit |
| 14:33.54 | ck_28 | asterisk 1.6.0.9 32 bit |
| 14:34.45 | Chainsaw | Could you confirm with uname -a output please? |
| 14:34.59 | Chainsaw | I'd hate to make a suggestion that won't work for you, just because I don't have the full picture. |
| 14:35.08 | ck_28 | Linux pbx.local 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux |
| 14:35.52 | ck_28 | linux 32 bit |
| 14:37.13 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 14:37.34 | ck_28 | Chainsaw how can i know the version of g729 i should get ? |
| 14:37.46 | Chainsaw | Agreed, i686. |
| 14:38.00 | Chainsaw | The one you mentioned seemed right to me, provided your system is compiled with GCC 4. |
| 14:38.08 | Chainsaw | Can I have the first line of output for gcc -version please? |
| 14:38.22 | Chainsaw | gcc --version, even. Just the first line, not the whole blurb. |
| 14:38.39 | ck_28 | Chainsaw how can i know the version of g729 i should get ? |
| 14:38.51 | Chainsaw | Please answer the question first. |
| 14:38.55 | ck_28 | gcc (GCC) 4.1.2 20071124 (Red Hat 4.1.2-42) |
| 14:39.03 | ck_28 | sorry man bad copy and paste |
| 14:39.17 | Chainsaw | Okay, then the version you mentioned there, codec_g729-ast16-gcc4-glibc-pentium4.so, seems appropriate. |
| 14:39.31 | Chainsaw | You have a Pentium IV-based Xeon CPU, are using GCC4 & glibc. |
| 14:39.43 | Chainsaw | The only variable is what Asterisk version you're using, as ast16 to me says Asterisk 1.6 |
| 14:39.49 | tegg | how can i check that the HFC Card is configured ? |
| 14:40.12 | ck_28 | asterisk 1.6 |
| 14:40.28 | ck_28 | Asterisk 1.6.0.6 |
| 14:41.20 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 14:43.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:46.31 | Chainsaw | ck_28: That seems to me like the correct .so for your system then. |
| 14:46.55 | joesuffceren | blast. I rebooted the pbx, and the problem was resolved for 10 minutes, but now it's back. This problem causes/correlates with high CPU usage on my box if that helps |
| 14:49.13 | *** join/#asterisk arpu (n=arpu@88-117-12-249.adsl.highway.telekom.at) |
| 14:49.39 | doolittlework | sorry joesuffceren did u come right had to step out for awhile? |
| 14:50.13 | ck_28 | Chainsaw it is not working if i allow g729 and disallowed all |
| 14:50.15 | joesuffceren | no, still having issues. I had corrected it temporarily with a reboot |
| 14:50.47 | ZefK | Hi, I saw an old information (http://lists.digium.com/pipermail/asterisk-dev/2003-September/001751.html) that there is a patch for libpri to display the UUI field. Is it possible to access the UUI info from dialplan? Thanks. |
| 14:50.59 | Chainsaw | ck_28: And the other end supports g729? |
| 14:51.19 | Chainsaw | ck_28: It's one of those codecs that tends to require a license, so it defaults to off in most devices I've seen. |
| 14:51.39 | doolittlework | must i give a copy of my configs? to check it out? |
| 14:53.15 | joesuffceren | doolittlework: was that last at me? If so, I'm sorry, but I don't follow |
| 14:53.56 | *** part/#asterisk thuglife (i=robert@blade2k.humppa.hu) |
| 14:54.49 | doolittlework | yip getig configs for u |
| 14:56.02 | [TK]D-Fender | ck_28: Show us the failed call with SIP debug enabled |
| 14:56.04 | [TK]D-Fender | ~pb |
| 14:56.05 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 14:56.07 | [TK]D-Fender | ^^^^^^ |
| 14:56.33 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:57.49 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:58.10 | doolittlework | joesuffceren: try adding : pridialplan=unknown and prilocaldialplan=unknown in zapata.conf |
| 15:00.09 | *** join/#asterisk alerios (n=alerios@190.144.75.22) |
| 15:03.57 | *** join/#asterisk Attilas (n=lord_att@modemcable048.42-83-70.mc.videotron.ca) |
| 15:07.49 | *** join/#asterisk Marquel (n=Flinx@port-7589.pppoe.wtnet.de) |
| 15:09.07 | Katty | :> |
| 15:09.21 | Katty | :>>> |
| 15:09.23 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 15:09.27 | Katty | boingboingboing |
| 15:10.09 | [TK]D-Fender | Katty: Mew. |
| 15:10.37 | Katty | [TK]D-Fender: mew! |
| 15:11.12 | eppigy | hello Katty |
| 15:11.19 | eppigy | ^______________^ |
| 15:11.41 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 15:11.43 | Katty | hello, dave. |
| 15:11.56 | Katty | has reached optimum caffeination. |
| 15:12.20 | eppigy | :D |
| 15:12.23 | eppigy | i am almost there |
| 15:12.42 | Katty | :> |
| 15:18.00 | MaliutaLap | mmm caffeine |
| 15:18.12 | MaliutaLap | although right now it would be a bad idea |
| 15:19.18 | Attilas | Just want to say Hello to all of you ! Almost 10 year since I logged in MIRC :) |
| 15:19.39 | MaliutaLap | we don't care about your client |
| 15:20.21 | Katty | maybe we don't care about you |
| 15:20.25 | Attilas | Ohh yeah should say IRC :) |
| 15:20.25 | Katty | mister poor attitude MaliutaLap |
| 15:20.45 | Attilas | Katty: Allright, just wanted to be friendly... |
| 15:20.59 | MaliutaLap | Katty: you can take my GVHD then |
| 15:21.11 | Katty | i've got yer GVHD right here |
| 15:21.45 | MaliutaLap | well considering it _was_ an american woman who donated the marrow ... you may very well do |
| 15:22.21 | MaliutaLap | you're not a 55y/o native american woman who donated bone marrow 4.5 years ago, are you? |
| 15:22.27 | Katty | YES |
| 15:22.29 | Katty | YES IA M |
| 15:22.48 | MaliutaLap | stabs Katty |
| 15:23.01 | MaliutaLap | donor immune system si a PITA |
| 15:23.34 | Katty | you're a PITA |
| 15:24.42 | Katty | oh come on. |
| 15:24.45 | Katty | play the game. |
| 15:24.49 | Katty | you're no fun. |
| 15:24.51 | Katty | eppigy: PITA. |
| 15:25.06 | MaliutaLap | the game is stab the donor right now |
| 15:25.23 | Katty | i've been stabbed enough for one lifetime |
| 15:25.23 | MaliutaLap | after 2 days of pain ... almost 5 years on I'm sick of it |
| 15:25.52 | Katty | sitting around dying in ER you should only have to do once in your life |
| 15:25.59 | Katty | but lucky me will get to do it again someday i'm sure! |
| 15:26.13 | MaliutaLap | fecking flare means I'm at least one more month off being rid of the pred |
| 15:26.29 | MaliutaLap | Katty: been there, done that ... at least twice |
| 15:26.37 | *** join/#asterisk arpu (n=arpu@chello080109017107.12.14.vie.surfer.at) |
| 15:27.01 | MaliutaLap | Katty: and it will happen again ... numbers say the ALL will get me before 60 |
| 15:27.05 | Katty | wel isn't that just exciting! |
| 15:30.01 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 15:30.15 | eppigy | PITA? D: |
| 15:30.22 | Katty | people always take my random trolling the wrong way |
| 15:30.26 | Z-Row | Hi all. Does anyone have any pointers or links for func_odbc troubleshooting? All my read= queries are returning empty sets. Asterisk 1.4 on CentOS 5.1 with a local postgres db. The same queries work when run via `echo "<query>" | isql -v asterisk-account`. |
| 15:30.27 | Katty | eppigy: yes. PITA. |
| 15:30.31 | Katty | eppigy: then you say, YOU"RE A PITA |
| 15:30.37 | Katty | eppigy: and then i say, YOUR FACE IS A PITA |
| 15:30.39 | Katty | eppigy: etc. |
| 15:30.44 | eppigy | o |
| 15:30.52 | eppigy | you a etc |
| 15:31.01 | Katty | your face is a etc |
| 15:31.07 | Z-Row | "You're a towel!" |
| 15:31.08 | Katty | (an?) |
| 15:31.30 | eppigy | youre some parenthesis |
| 15:31.36 | Katty | your mom is. |
| 15:31.38 | eppigy | with an a and n |
| 15:31.45 | Katty | wel. |
| 15:31.46 | Katty | well. |
| 15:31.49 | Katty | YOUR FACE. |
| 15:31.52 | MaliutaLap | no, I'm Hoopy ... I'm a frood who really knows where his towel is |
| 15:32.03 | eppigy | MY FACE HAS THE FINE LINES OF A LOWERCASE A YES |
| 15:32.11 | eppigy | YEAH |
| 15:32.22 | Katty | that did not parse. |
| 15:32.33 | Katty | i think my parses broken. |
| 15:32.37 | Katty | parser. |
| 15:32.41 | eppigy | :< |
| 15:32.53 | Katty | :< |
| 15:33.03 | Katty | so how about some lunch. |
| 15:33.11 | Attilas | Hmmm is this channel support for Asterisk PBX or .... ? |
| 15:33.16 | Katty | no. |
| 15:33.18 | Katty | no it's not. |
| 15:33.22 | Attilas | damn :( |
| 15:33.24 | MaliutaLap | sometimes |
| 15:33.26 | Attilas | seriously ? |
| 15:33.28 | Attilas | lol |
| 15:33.28 | eppigy | I am hungry |
| 15:33.34 | Katty | Me too. |
| 15:33.36 | jaytee | no, this is a channel for discussing homemade cured meats |
| 15:33.36 | MaliutaLap | when we feel like it |
| 15:33.42 | eppigy | meats |
| 15:33.44 | [TK]D-Fender | eppigy: You can't be hungry, you said you were dave!!!! |
| 15:33.45 | Katty | Fancy a grilled cheese with some tomato soup? |
| 15:33.48 | eppigy | D: |
| 15:33.49 | [TK]D-Fender | eppigy: LIES |
| 15:33.53 | eppigy | CAUGHT IN A LIE |
| 15:34.00 | eppigy | COMMENCE BACKPEDDLING |
| 15:34.02 | Katty | caught in a web of lies. |
| 15:34.05 | eppigy | HUNGRY IS MY MAIDEN NAME |
| 15:34.05 | Katty | by [TK]D-Fender. |
| 15:34.06 | MaliutaLap | jaytee: I thought it was for trading desert recipes |
| 15:34.07 | Katty | the spider. |
| 15:34.22 | Katty | who needs desserts when you have carbs and cheese. |
| 15:34.34 | eppigy | Katty is so sweet who needs dessert |
| 15:34.40 | [TK]D-Fender | eppigy>HUNGRY IS MY MAIDEN NAME <- you're anything but "fair" :p |
| 15:34.48 | Katty | haha |
| 15:34.48 | MaliutaLap | me, I need my sticky sweet tart tartan |
| 15:34.52 | eppigy | 8[] |
| 15:34.54 | Katty | awards [TK]D-Fender 1 point. |
| 15:34.56 | eppigy | how dare you |
| 15:35.03 | jaytee | everytime I tell someone how I make "The World's Greatest Grilled Cheese" I get odd looks and doubtful comments......until they actually try one and then I've hooked them. |
| 15:35.04 | MaliutaLap | [TK]D-Fender: you keep a maiden? |
| 15:35.05 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:35.05 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:35.15 | Katty | jaytee: :> |
| 15:35.21 | Katty | jaytee: marbled rye and smoked mozzarella? :> |
| 15:35.25 | jaytee | nope |
| 15:35.29 | Katty | :< |
| 15:35.36 | Katty | details. |
| 15:35.42 | eppigy | SPILL ITY |
| 15:35.45 | jaytee | I use white Land O' Lakes American cheese |
| 15:35.46 | [TK]D-Fender | MaliutaLap: Only a harem ;) |
| 15:35.55 | jaytee | I use whole wheat bread |
| 15:35.57 | Katty | never heard of it. |
| 15:36.00 | [TK]D-Fender | AMERICAN CHEESE. Dear God... |
| 15:36.10 | Katty | [TK]D-Fender: SHUUUSH! this is jaytee's story. |
| 15:36.13 | MaliutaLap | [TK]D-Fender: a harem is actually a building, the section of the house where the women live |
| 15:37.25 | eppigy | well I am glad I live in the 18th century |
| 15:37.32 | eppigy | so that knowledge will be applicable |
| 15:37.48 | jaytee | I spread a very thin layer of spicy brown mustard on one side of a piece of wheat bread. I put two pieces of cheese on the mustard side of the bread and top with another slice of bread. I spread a very, very thin layer of Hellman's mayo on the outside of the sandwich and cook to a golden brown over medium low heat on a non-stick griddle with no oil or butter. |
| 15:38.19 | eppigy | oh boi |
| 15:38.26 | MaliutaLap | jaytee: coco van? |
| 15:38.40 | Katty | jaytee: mayo? |
| 15:38.41 | MaliutaLap | jaytee: it's not a coq missure |
| 15:38.44 | Katty | jaytee: in place of the butter? |
| 15:38.45 | jaytee | coq au vin? |
| 15:38.54 | Katty | fascinating. |
| 15:38.58 | MaliutaLap | Katty: it's an old french recipe |
| 15:39.01 | jaytee | mayo, in a very thin layer in place of the butter |
| 15:39.19 | Katty | neat. |
| 15:39.44 | jaytee | it gives it a rich golden brown crust like when you brush egg whites on something you're baking to give it a golden brown glaze |
| 15:40.13 | [TK]D-Fender | [11:38]<MaliutaLap>jaytee: it's not a coq missure <- pardon? |
| 15:40.24 | jaytee | but this way the grilled cheese isn't all greasy with butter (which some people actually prefer) |
| 15:40.50 | *** join/#asterisk af_ (n=getsmart@88-149-230-50.dynamic.ngi.it) |
| 15:41.58 | jaytee | I've actually used a little bit of whipping cream in my scrambled egg mix in order to make them more "fluffy" |
| 15:42.08 | MaliutaLap | [TK]D-Fender: there is a name for it, it's a french sandwich. From memory that's the name I came up with |
| 15:44.06 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:44.31 | MaliutaLap | Croque-monsieur |
| 15:45.08 | Attilas | Croque monsieur is made with tomatoes, egg and cheese |
| 15:45.08 | [TK]D-Fender | MaliutaLap: Better :) |
| 15:45.10 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:45.33 | Attilas | They are toested to... so not really fluffy :) |
| 15:45.50 | Attilas | <--- French Frog :) |
| 15:45.57 | MaliutaLap | Attilas: none of the recipes I have seen do that |
| 15:46.15 | [TK]D-Fender | <--- English Frog |
| 15:46.53 | eppigy | TRABAJO |
| 15:47.13 | MaliutaLap | Attilas: from what I know you are talking about a variation done in provincal |
| 15:47.17 | [TK]D-Fender | eppigy: Va t'ens tabarnac! |
| 15:47.21 | [TK]D-Fender | :D |
| 15:47.27 | MaliutaLap | <---- foodie |
| 15:50.11 | jchillerup | has got his RickRoll asterisk setup working |
| 15:50.14 | jchillerup | \o/ |
| 15:52.05 | *** part/#asterisk alerios (n=alerios@190.144.75.22) |
| 15:52.05 | eppigy | GREAT |
| 15:55.11 | jchillerup | :) |
| 15:57.34 | russellb | may or may not have a rickroll.ulaw and may or may not have injecting that into random conference bridges ... |
| 15:57.45 | russellb | s/injecting/injected/ |
| 15:58.35 | coppice | rickroll might be one of the few things which can't sound worse in narrowband form |
| 15:59.00 | russellb | :-) |
| 16:00.40 | *** join/#asterisk shareenergy (n=go@host-87-74-7-50.dslgb.com) |
| 16:00.47 | shareenergy | Hello guys |
| 16:01.04 | shareenergy | Is any way of call recording only specif agents? |
| 16:01.12 | therealcircut | tts == teh suck |
| 16:01.39 | jchillerup | I have some problems, though, because it doesn't always answer the phone |
| 16:01.46 | jchillerup | the calls* |
| 16:01.52 | jchillerup | It's on SIP |
| 16:02.41 | [TK]D-Fender | shareenergy: Do your own monitor in the dialplan running those agents as local channels |
| 16:03.04 | nephfl | hello, I'm back at that same problem with the call files with the TDM card connecting internal extensions to an app |
| 16:03.22 | nephfl | anyone have any suggestions as to why it is buggy? |
| 16:03.43 | [TK]D-Fender | nephfl: Works fine for the rest of us, maybe you should show us your problem. |
| 16:04.19 | nephfl | im not sure what to show, sometimes it will ring, other times it changes the status of the channel to ringing, but the phone doesnt ring |
| 16:04.48 | nephfl | sometimes it says it was picked up |
| 16:05.27 | [TK]D-Fender | nephfl: Yeah, but you NEVER show us anything, and that's consistent! |
| 16:05.29 | *** join/#asterisk stijnbe (n=stijnbe@78-22-110-114.access.telenet.be) |
| 16:05.31 | [TK]D-Fender | ;) |
| 16:05.40 | [TK]D-Fender | nephfl: Pastebin it! |
| 16:07.05 | nephfl | ok |
| 16:07.21 | nephfl | the only info i get from cli is -- Attempting call on Zap/3-1 for application VoicemailMain() (Retry 1) |
| 16:08.05 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 16:08.28 | jchillerup | We're no strangers to looooove |
| 16:08.33 | jchillerup | you know the rules, and so do I |
| 16:13.16 | nephfl | http://pastebin.com/d173a51e4 |
| 16:14.12 | [TK]D-Fender | nephfl: Channel: Zap/3-1 <- not legal |
| 16:14.27 | [TK]D-Fender | nephfl: that is a call channel, but not a legal device to call. |
| 16:14.34 | [TK]D-Fender | nephfl: Drop the "-1" |
| 16:14.57 | nephfl | i added the -1 because it did nothing..let me make the change and see what happnes |
| 16:15.45 | nephfl | <PROTECTED> |
| 16:17.27 | [TK]D-Fender | nephfl: PB the actual attempt with a chnnel dump prior |
| 16:17.57 | nephfl | dont know how to do channel dump |
| 16:18.24 | [TK]D-Fender | nephfl: core show channels concise |
| 16:19.01 | *** join/#asterisk Flyser (n=Flyser@unaffiliated/flyser) |
| 16:20.19 | nephfl | just get -- Attempting call on Zap/3 for application VoicemailMain() (Retry 1) |
| 16:20.27 | EricDGlobeTroote | hi guys,, im getting this error on my box.. does anyone know how to fix this |
| 16:20.28 | EricDGlobeTroote | frame.c:145 smoother_frame_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 16:20.29 | nephfl | and -- Hungup 'Zap/3-1' |
| 16:21.37 | *** join/#asterisk frantic667 (n=toffifee@dsbg-4db5cec1.pool.einsundeins.de) |
| 16:21.43 | frantic667 | hi there |
| 16:21.50 | eppigy | TRANAJO |
| 16:21.55 | eppigy | TRABAJO |
| 16:22.32 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 16:22.39 | nephfl | no other info from CLI |
| 16:22.52 | joesuffceren | [TK]D-Fender: my incoming call issue seems to be temporarily resolved after I powered down the machine, removed it's power cable, attempted to power it on (to clear all power from PS and MB) then cabled back up and powered back up. I fear it may come again, though (has recurred 4 times in two weeks). any thoughts on further troubleshooting? |
| 16:23.02 | frantic667 | i have still a problem with my provider (german, provider is 1und1), he uses several sip gateways, asterisk doesn't recognise a call... outgoing works, but no incoming |
| 16:23.30 | [TK]D-Fender | joesuffceren: No other details to go on... |
| 16:25.20 | frantic667 | has someone a working config? a few days ago some calls were routed correctly, but today 0% of my calls reach the asterisk. not one character output in the CLI, any idea? is it possible that the provider blocks my account? |
| 16:25.54 | *** join/#asterisk Alborracho (n=chatzill@190.25.135.1) |
| 16:26.04 | joesuffceren | [TK]D-Fender: I'll be happy to provide any details you want if there's anything that can help. any possibility it could be a hardware issue with my PRI card? |
| 16:26.09 | Alborracho | hi everyone |
| 16:26.25 | [TK]D-Fender | frantic667: Intermittent incoming is a networking issue if you are not getting the SIP request at all |
| 16:26.38 | [TK]D-Fender | frantic667: Most common is improper NAT setup |
| 16:26.45 | [TK]D-Fender | frantic667: Is your server behind NAT? |
| 16:26.50 | *** join/#asterisk mvanbaak_ (i=mvanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 16:27.37 | *** join/#asterisk ctp (n=ctp@brsg-d9bee524.pool.mediaWays.net) |
| 16:27.45 | frantic667 | yes, it's behind a NAT. externip is set to my dyndns account and the server is set to "exposed host" |
| 16:28.02 | [TK]D-Fender | frantic667: Takes a lot more than that. Go read the guide : |
| 16:28.04 | [TK]D-Fender | ~sipnat |
| 16:28.05 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:28.06 | [TK]D-Fender | ^^^^^ |
| 16:28.50 | frantic667 | okay, thanks a lot, i'll walk through that and come back if I still need help ;-) |
| 16:30.27 | nephfl | any other ideas how i can determine the reason my call files arent working? the extensions work dialing each other or applications..just dont ring for the call file |
| 16:31.48 | [TK]D-Fender | nephfl: I jsut asked for a complete attempt visible from CLI witha channel dump... |
| 16:32.07 | nephfl | I pasted it to this window, everything the cli output anyway |
| 16:32.14 | Alborracho | can anyone help me with a g729 dtmf issue? |
| 16:32.31 | nephfl | ill try again and pastebin |
| 16:35.06 | nephfl | http://pastebin.com/d4861e090 |
| 16:39.47 | [TK]D-Fender | nephfl: No ring ro anything? |
| 16:40.07 | [TK]D-Fender | nephfl: I don't see a ring or pickup notice, or anything much really. |
| 16:40.11 | nephfl | no it has rang a few times, but no |
| 16:40.35 | nephfl | I know, thats why i have no idea what to do about it |
| 16:41.00 | nephfl | i dont think anything was different the few times it did work |
| 16:41.46 | [TK]D-Fender | nephfl: are you picking up? |
| 16:42.17 | nephfl | i just did line 4 and it worked |
| 16:42.38 | [TK]D-Fender | nephfl: repeatedly? |
| 16:42.55 | nephfl | hmm channel 4 seems to be working |
| 16:43.28 | nephfl | right now its just channel 3 not working |
| 16:43.36 | nephfl | maybe i should reseat the daughter board |
| 16:44.09 | [TK]D-Fender | nephfl: A fine idea once you've done a few more tests against other channels. |
| 16:44.14 | *** join/#asterisk RoadKill (n=Miranda@gate.ipvision.dk) |
| 16:44.32 | [TK]D-Fender | nephfl: Then SWAP modules and test to see if its the port, then try the other channel to see if its the module |
| 16:44.40 | [TK]D-Fender | nephfl: multiple tests each |
| 16:45.07 | nephfl | sounds like a plan |
| 16:55.26 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 16:57.31 | *** join/#asterisk itess (n=kvirc@85.17.174.172) |
| 16:58.30 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:01.12 | itess | hello. can someone help me with configuring phone for asterisk? the phone is speedport 503W -- dsl router with phone jack. the phone works well with external sip provider, but i can't make it to work with my local asterisk. |
| 17:01.40 | itess | here is phone config: http://pastebin.ca/1427617 |
| 17:02.52 | therealcircut | anyone got fax to email working under asterisk 1.4.x? |
| 17:02.59 | itess | the problem looks strange: phone registers itself with asterisk, but doesn't use it when i am dialing |
| 17:03.42 | itess | it just says "busy" |
| 17:04.56 | *** join/#asterisk lucasb (n=bussey@s154-5-252-231.bc.hsia.telus.net) |
| 17:05.32 | [TK]D-Fender | itess: shat says "busy"? |
| 17:05.39 | itess | phne |
| 17:05.50 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 17:06.42 | [TK]D-Fender | itess: What does * say with SIP DEBUG enabled? |
| 17:07.01 | itess | nothing. only register and notify commands |
| 17:07.16 | ruben23 | hi |
| 17:07.18 | [TK]D-Fender | itess: then your phone is misconfigured |
| 17:07.47 | ruben23 | how do i check if my TDM400P is installed and working a my timing device for my asterisk server.. |
| 17:08.03 | itess | [TK]D-Fender: probably. it has only basics sip settings, but it works with voicetrading |
| 17:08.21 | [TK]D-Fender | ruben23: Remove dahdi_dummy and see if meetme works |
| 17:08.35 | [TK]D-Fender | itess: Most devices loike that are locked to their provider |
| 17:09.05 | itess | [TK]D-Fender: what exactly can be wrong? auth works, phone registers itself, * sends notifies without problem |
| 17:09.21 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-dd7d48cd4c3c3fd6) |
| 17:09.37 | itess | well, it doesn't look like this. |
| 17:09.46 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-5def4dd3d4374534) |
| 17:09.46 | [TK]D-Fender | itess: Well clearly the phone is not sending the call to * so its the devices fault |
| 17:10.17 | itess | yeah, looks so |
| 17:11.11 | itess | [TK]D-Fender: i would think the problem is that asterisk is behind the router, not before it. can this be the case? |
| 17:11.58 | [TK]D-Fender | itess: the devices registers to * in the first place. No reason 1 part of functionailty would work in that direction and not the other |
| 17:12.21 | *** join/#asterisk s0lid (n=s0lid@122.53.97.103) |
| 17:12.34 | [TK]D-Fender | itess: Keeping in mind that registration and call servers CAN be different, we don't see your configs, and we don't know what it might be hiding from you as well. |
| 17:12.50 | *** join/#asterisk Dovid (n=annon@ool-4355e297.dyn.optonline.net) |
| 17:13.11 | itess | [TK]D-Fender: it's difficult to describe short, as i am using vpn :) |
| 17:13.27 | [TK]D-Fender | itess: Still falls under the first issue |
| 17:13.34 | Dovid | hi. I believe i know the answer to this quesiton but the boss wants me to ask any ways. is there any way that if a call is under X amount of seconds to wait till Y seconds before hanging up even though the use hung up ? |
| 17:13.45 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 17:13.46 | [TK]D-Fender | itess: If reg works, you should at least SEE calls, unless the device si misconfigured somewhere else |
| 17:14.14 | Dovid | TK: If you answer no to my question u will make life a lot easier ;) |
| 17:14.42 | [TK]D-Fender | Dovid: Wait Y seconds for who? |
| 17:16.02 | Dovid | TK: SNOM -> Asterisk -> Carrier. carrier has a penalty if call is answered and it is under 6 seconds. we dont want that penalty. cheaper to send the hangup after 6 seconds then send the hangup before the 6 second mark |
| 17:16.14 | tegg | hi, is asterisk starting when there is an problem with nat ? cause i read that i must open some ports for asterisk, but i didn't do this since |
| 17:16.37 | [TK]D-Fender | ~sipnat |
| 17:16.38 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:16.40 | [TK]D-Fender | tegg: ^^^^^^^ |
| 17:16.44 | Dovid | tegg: Asterisk will start. if you are behind nat then u need to forward ports to make it work ;) |
| 17:17.06 | [TK]D-Fender | Dovid: So when YOUR phone hangs up in < 6s you want to extend the call... |
| 17:17.12 | Dovid | correct |
| 17:17.23 | [TK]D-Fender | Dovid: I'm sure there's a way, though I'm thinink tricky... |
| 17:17.26 | Dovid | didnt think there was a way to do it in asterisk but i was told to ask so i am |
| 17:17.28 | [TK]D-Fender | thinks on it |
| 17:17.34 | tegg | ok, than this is not my problem :) |
| 17:17.34 | Dovid | TK: I was thinking of just patching. |
| 17:17.47 | [TK]D-Fender | Dovid: thats insane |
| 17:17.47 | tegg | 'cause asterisk won't start |
| 17:17.47 | Dovid | tegg: What issues r u having when asterisk starts ? |
| 17:17.50 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 17:18.06 | Dovid | TK: How would you do it ? |
| 17:18.07 | tegg | it stop with "asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_esel.so: undefined symbol: ast_extension_state_callerid_add |
| 17:18.07 | tegg | " |
| 17:18.12 | [TK]D-Fender | tegg: And I see no elaboration on "doesn't start' with maybe some kind of error, ect.. |
| 17:18.24 | tegg | mom |
| 17:19.13 | [TK]D-Fender | tegg: if i were you I'd NOLOAD that one in modules.conf to see if its the culprit as it might appear. |
| 17:19.26 | [TK]D-Fender | tegg: file already told you that isn't a standard module. |
| 17:20.04 | tegg | but nobody told me that i can >>noload<< this :) |
| 17:20.19 | Dovid | tegg: Read the book. what else should we have told u ? |
| 17:20.20 | Dovid | !book |
| 17:20.23 | Dovid | ~book |
| 17:20.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:22.38 | *** join/#asterisk jarg (n=jarg@201.155.146.208) |
| 17:22.40 | Dovid | TK: Any idea on how I would do it ? |
| 17:22.59 | [TK]D-Fender | tegg: Well I just did. Merry Christmas. |
| 17:23.09 | [TK]D-Fender | Dovid: still thinking on it |
| 17:23.13 | Dovid | ok. |
| 17:23.51 | jameswf | tzafrir_laptop: ping |
| 17:24.13 | tzafrir_laptop | jameswf, pong |
| 17:24.16 | Dovid | lol |
| 17:25.17 | [TK]D-Fender | smashes jameswf deep into tzafrir_laptop's back court FTW! |
| 17:25.57 | Dovid | tk: sent u a pm |
| 17:26.16 | jameswf | tzafrir_laptop: FYI our digital cards are not in Chans.pm... When I add it i get unknown default dahdi signalling for chan 1 type PRI at /usr/lib/perl5/site_perl/5.8.8/Dahdi/Config/Gen/System.pm line 90. |
| 17:26.16 | [TK]D-Fender | Dovid: Promptly closed the tab, no reason to take it to PM :) |
| 17:26.55 | tzafrir_laptop | jameswf, is it in latest SVN? |
| 17:27.01 | tzafrir_laptop | (dahdi linux) |
| 17:27.29 | tzafrir_laptop | dahdi-tools, that is |
| 17:28.31 | jarg | hello all, i have a question about "Fax for Asterisk" some one know if i need a license per sip channel and dahdi channel? |
| 17:28.45 | [TK]D-Fender | jarg: Per channel PROCEssING a fax |
| 17:29.01 | jameswf | tzafrir_laptop: okay I see it, thoughts on the PRI error? |
| 17:29.38 | tzafrir_laptop | jameswf, do you still get the error with latest dahdi-tools svn? if so: what exactly? |
| 17:30.14 | tzafrir_laptop | jameswf, no need to actually install it. |
| 17:30.37 | tzafrir_laptop | just run: ./xpp/dahdi_genconf # from a fresh checkout |
| 17:31.02 | jameswf | yeah I am on a clean box so I have to install subversion etc... brb |
| 17:32.41 | tzafrir_laptop | or better: DAHDI_CONF_FILE=/tmp/system.conf ./xpp/dahdi_genconf system |
| 17:35.06 | *** join/#asterisk lasko (n=lasko@70.99.232.106) |
| 17:35.09 | *** join/#asterisk rhassing (n=rob@redcap.xs4all.nl) |
| 17:36.06 | jarg | [TK]D-Fender, thanks, i have 3 multifunctional machines which are used for send fax this have analog ports, i can connect to my * with a ATA as SIP peers and send fax using Fax for Asterisk? in this scenario i only need 3 licenses? |
| 17:36.40 | [TK]D-Fender | jarg: I'm not sure how many different ways we can say the same thing to you. |
| 17:36.42 | [TK]D-Fender | jarg: PER SIMULTANEOUS CALL |
| 17:36.57 | [TK]D-Fender | jarg: # little words. It ain't raw-Cat Science. |
| 17:37.00 | jameswf | tzafrir_laptop: all is well @ trunk... |
| 17:37.47 | coppice | jarg: if you are just trying to get your FAX machines to talk to the outside world, why would you need a FAX engine in your PBX? |
| 17:38.40 | tzafrir_laptop | coppice, unrelated issue: what version of libsupertone works with current spandsp? |
| 17:39.14 | coppice | tzafrir_laptop: according to your report, none. I need to fix that |
| 17:39.34 | jarg | [TK]D-Fender, ok, i understand now, thanks :P |
| 17:42.27 | jarg | coppice, really my server also would be recive fax :P |
| 17:43.09 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
| 17:43.17 | paulius | Any SPA-3102 fanboys in here? |
| 17:43.32 | coppice | no. SPA3102s suck |
| 17:43.52 | paulius | last time I was here people were saying that I suck for not loving the 3102 |
| 17:44.18 | coppice | well, most other ATAs suck even more than the SPA3102 |
| 17:44.50 | paulius | well at least it has that, then. |
| 17:45.00 | [TK]D-Fender | DISCLAIMER : coppice believes that just about all telecom related gear sucks, especially anything the common man has access to. |
| 17:45.26 | [TK]D-Fender | paulius: No, you just suck :) |
| 17:45.43 | coppice | the T.38 in the SPA3102 is flaky, and lots of people report high hardware failure rates. apart from that, they're great |
| 17:45.46 | [TK]D-Fender | paulius: Possessing a fairly decent ATA like that is merely accessorizing ;) |
| 17:45.47 | paulius | I have almost the same hypothesis for networking gear, but I'm regarding telcom gear as neutral as I havn't had the chance to actually play with it. |
| 17:47.52 | *** join/#asterisk natmlt (n=Adium@ip68-3-162-244.ph.ph.cox.net) |
| 17:49.29 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-ebeedab7061a5720) |
| 17:49.31 | [TK]D-Fender | coppice: I've heard some specific reports that CPU load from T.38 is often responsible for early Linksys deaths... this what you're referring to? |
| 17:49.47 | [TK]D-Fender | coppice: Because I haven't heard anything about general spontaneous failures. |
| 17:49.50 | paulius | Hey, if I may interject, what is T.38? |
| 17:50.08 | paulius | And aren't those "deaths" in crappy consumer hardware usually linked to bad capacitors? |
| 17:50.11 | [TK]D-Fender | paulius: If you have to ask, you really don't need to. |
| 17:50.15 | paulius | hehe okay |
| 17:50.26 | [TK]D-Fender | paulius: be thankful |
| 17:50.56 | paulius | Hey great. |
| 17:51.04 | paulius | Now it thinks that the WAN access option is invalid |
| 17:51.11 | coppice | no. people not using T.38 report lots of failures too. the T.38 implementation sucks. its really fussy about the fax machine you use. I've changed the length of the lead from the fax to the SPA3102 and gone from lots of failures to none. its that quirky |
| 17:51.16 | [TK]D-Fender | Oxymoron : See "Interoperability" |
| 17:52.35 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:53.14 | bmoraca | just set the speed of the fax machine to 9600 and you won't have any problems :P |
| 17:53.24 | *** join/#asterisk chainsawf (n=chainsaw@121-73-114-254.cable.telstraclear.net) |
| 17:53.33 | paulius | The first thing that I can't stand about the 3102 is the GUI. |
| 17:53.41 | paulius | There's even no save settings to file option! |
| 17:53.51 | paulius | And the guy doesn't look anything like any of the other Linksys gear. |
| 17:54.00 | paulius | Makes it look like some cheap rebrand from some no-name chinese device. |
| 17:54.04 | *** join/#asterisk stijnbe (n=stijnbe@78-22-110-114.access.telenet.be) |
| 17:55.00 | coppice | it looks exactly like the other linksys VoIP stuff |
| 17:55.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:55.39 | voxter | yeah i dont know what you're talking about, it looks like every other piece of linksys hardware |
| 17:55.41 | paulius | I havn't had the "chance" to use that. |
| 17:55.48 | [TK]D-Fender | paulius: Its based on Sipura's model which was made by ex-Cisco techs |
| 17:55.50 | bmoraca | any reason to use the SPA3102 instead of the PAP2T? is it more powerful or somethinbg? |
| 17:56.02 | [TK]D-Fender | paulius: And you can provision them, you do not EXPORT GUI settings to file. |
| 17:56.10 | paulius | [TK]D-Fender: Ah. |
| 17:56.14 | [TK]D-Fender | paulius: You have a reverse sense of direction here... |
| 17:57.59 | paulius | [TK]D-Fender: So I want to get my echo problems solved. I've just reset my router... I had a few people recommend me some tutorials in here or #freepbx |
| 17:58.20 | paulius | I've got this page (http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx) and then this one (http://www.voipedia.pl/index.php/SPA_3102_i_echo) |
| 17:58.42 | paulius | Someone (maybe even you) mentionned that the second one has better settings for the echo. |
| 17:59.02 | coppice | the EC in the SPA3102 has "issues" |
| 17:59.07 | paulius | Should I follow the first page and ignore any echo settings |
| 17:59.32 | paulius | coppice: Yeah but I don't have "issues", I have big echo. I'm sure that it's a configuration issue. |
| 18:00.09 | coppice | maybe, but some people really do have big echos using them when things are configured properly |
| 18:00.37 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) [NETSPLIT VICTIM] |
| 18:00.37 | *** join/#asterisk foo (n=foo@unaffiliated/foo) [NETSPLIT VICTIM] |
| 18:00.38 | *** join/#asterisk rbd (n=rbd@rrcs-96-10-27-206.se.biz.rr.com) [NETSPLIT VICTIM] |
| 18:00.39 | *** join/#asterisk Corydon76-dig (i=three@pdpc/supporter/bronze/Corydon76-home) [NETSPLIT VICTIM] |
| 18:00.39 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 18:00.57 | paulius | coppice: Well the other day people told me that I suck and that the 3102 should never have bad echo like I'm experiencing. |
| 18:01.31 | coppice | oh, a search for echo problems with the SPA3192 will get you lots of hits. |
| 18:01.57 | *** join/#asterisk cutlass (n=cutlass@c-67-165-132-67.hsd1.il.comcast.net) |
| 18:02.14 | [TK]D-Fender | paulius: and I directed you to voxilla's forums for this as well |
| 18:02.41 | cutlass | does anyone know if app_rpt can be configured to tear down the SIP session (BYE) when the 2 way radio talker stops transmitting? |
| 18:08.12 | eppigy | DONDE ESTA |
| 18:10.30 | telnettech | que? |
| 18:13.17 | _ShrikE | FRIJOLES |
| 18:15.04 | bmoraca | anyone have a recommendation for a good analog cordless phone? DECT would be preferred, but not necessary. prefer without built-in answering machine, would like MWI, and do NOT want Uniden... thanks :) |
| 18:15.40 | jameswf | writing perl I feel all dirty :x |
| 18:24.02 | *** join/#asterisk ctp (n=ctp@brsg-d9bee524.pool.mediaWays.net) |
| 18:24.45 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 18:26.34 | *** join/#asterisk plq (n=plq@88.250.169.4) |
| 18:28.52 | paulius | gmm great. |
| 18:29.00 | paulius | I think I got rid of the echo on outgoing calls |
| 18:29.13 | paulius | but now the local calls don't get dropped in the queue on my asterisk |
| 18:30.42 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:31.21 | b14ck | I'm not much of a perl fan, myself, jameswf :) |
| 18:31.35 | [TK]D-Fender | paulius: Wow, thats completely different. |
| 18:31.45 | paulius | yeah I got it solved though |
| 18:31.54 | paulius | Hey I have a dumb question. |
| 18:32.04 | paulius | What's the difference between Line 1 and PSTN Line? |
| 18:32.11 | paulius | And then they have the same for PSTN user and User 1 |
| 18:32.41 | jaytee | PSTN is the FXO (analog line to telco), Line 1 is the FXS |
| 18:32.54 | paulius | ah I see. |
| 18:35.29 | *** join/#asterisk nicola_pav (n=nick@bzq-79-180-241-137.red.bezeqint.net) |
| 18:36.11 | tzafrir_laptop | b14ck, you don't blow perl? |
| 18:36.20 | nicola_pav | hello. i have a linksys spa 3102 |
| 18:36.30 | nicola_pav | spa3102 has 1 fxs and i fxo |
| 18:36.37 | b14ck | I don't like perl much, but I've got a decent amount of experience with it. |
| 18:36.46 | nicola_pav | do i need to get both ports to register with asterisk? |
| 18:36.55 | tzafrir_laptop | paulius, someone for you :-) |
| 18:37.11 | paulius | hah |
| 18:37.24 | paulius | tzafrir_laptop: http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx |
| 18:37.35 | paulius | That's for the FXO port, the one that goes to the telco. |
| 18:37.36 | tzafrir_laptop | nicola_pav, --^ |
| 18:38.10 | nicola_pav | <PROTECTED> |
| 18:38.14 | tzafrir_laptop | nicola_pav, yes. it is technically two "sip phones", and both register with Asterisk |
| 18:38.29 | nicola_pav | another question please |
| 18:38.34 | paulius | tzafrir_laptop: Isn't one a trunk? |
| 18:38.48 | nicola_pav | i managed to register fxs using "host=dynamic" in sip.conf |
| 18:38.58 | tzafrir_laptop | paulius, well, yes |
| 18:39.01 | nicola_pav | can i do the same for the fxo? |
| 18:39.18 | nicola_pav | in other words, i don't want to use a fixed IP for the fxo |
| 18:39.32 | neurosys | How would you connect a channel bank to your asterisk box? |
| 18:39.50 | tzafrir_laptop | neurosys, what type of channel bank? |
| 18:41.05 | neurosys | tzafrir_laptop: Thats where Im stuck. I have a resort and they have 8 lines coming in. 28 phones in the resort. Im trying to figure out how I would connect the room phones to my asterisk box. |
| 18:42.02 | tzafrir_laptop | neurosys, what type of channel bank? E1? T1? SIP? tdmoe? |
| 18:45.23 | *** join/#asterisk tessier_ (n=treed@kernel-panic/sex-machines) |
| 18:45.27 | tessier_ | Hello all |
| 18:45.36 | neurosys | tzafrir_laptop: that's the thing. The 8 lines are coming in analog. Im thinking to run them into asterisk via FXOs. But how to i distribute to the rooms? |
| 18:46.57 | nicola_pav | let me put my question in another way |
| 18:47.31 | nicola_pav | can i do voip to pstn through the spa 3102 without giving fixed IP to the spa? |
| 18:47.37 | tzafrir_laptop | neurosys, again: how is the channel bank connected to asterisk? (what model and make is it, btw?) |
| 18:48.47 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 18:51.01 | *** part/#asterisk natmlt (n=Adium@ip68-3-162-244.ph.ph.cox.net) |
| 18:51.38 | *** join/#asterisk sikanrong (n=sikanron@8.104.219.87.dynamic.jazztel.es) |
| 18:52.03 | neurosys | tzafrir_laptop: sorry. I have not purchased one. Im trying to design out the whole phone system. Im under the impression i need a channel bank to connect the rooms to asterisk. Am i wrong? |
| 18:52.03 | sikanrong | I just got a voovox account and I'm trying to get asterisk to register the peer |
| 18:52.25 | sikanrong | everything in config looks okay, but in CLI when i do sip show registry i get nothing |
| 18:52.40 | sikanrong | is there something else I have to do to get asterisk to register? |
| 18:53.18 | sikanrong | i have the register line in the [general] section of sip.conf, it looks like this: |
| 18:53.33 | sikanrong | register=> user_id:secret_pass@eugw.ast.voovox.com |
| 18:53.59 | sikanrong | strangely, when i use a soft-phone to connect directly to the voovox account i can place calls just fine |
| 18:54.16 | sikanrong | anybody have any tips? |
| 18:54.18 | tzafrir_laptop | neurosys, as I'm involved with some specific type of channel bank vendor, I'll refrain fur further comments |
| 18:54.26 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:54.27 | [TK]D-Fender | nicola_pav: Yes |
| 18:54.31 | tzafrir_laptop | but generally, yes |
| 18:55.08 | [TK]D-Fender | neurosys: Distribute what to the rooms? |
| 18:55.37 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 18:55.51 | tzafrir_laptop | err.. hang on. 8 incoming fxo lines? |
| 18:55.55 | [TK]D-Fender | neurosys: AH... I'd really recommend a SIP gateway like Mediatrix 4100 series or AudioCodes Mediant. |
| 18:56.16 | tzafrir_laptop | you're in the US? |
| 18:56.17 | [TK]D-Fender | neurosys: PCI FXO card with HWEC, SIP gateway for the phones |
| 18:56.52 | nicola_pav | [TK]D-Fender: i manged to register the fxo using a fixed IP but could not do it using host = dynamic! |
| 18:56.57 | tzafrir_laptop | If you're in the US, maybe consider a fractional PRI line |
| 18:57.01 | nicola_pav | what i am missing? |
| 18:57.46 | [TK]D-Fender | nicola_pav: I can't tell what you're doing wrong so far. the SPA simply has to register to * which means * has to have the fixed hostname/IP |
| 18:58.10 | *** join/#asterisk lou_gr (n=louben@212-70-216-131.ath.static.tee.gr) |
| 18:59.00 | bmoraca | tzafrir_laptop: depends on where you are in the US...many places (California, for instance) won't do fractional PRIs unless you buy an integrated PRI, in which case you end up buying the whole T1 anyway |
| 18:59.58 | nicola_pav | [TK]D-Fender: to clear this out, registering fxo using host=dynamic is possible, right? |
| 19:01.08 | [TK]D-Fender | nicola_pav: host=dynamic is REQUIRED otherwise * refuses to allow registers |
| 19:02.06 | nicola_pav | in the case of the fxs i use host=dynamic |
| 19:02.15 | [TK]D-Fender | nicola_pav: use for both |
| 19:02.21 | nicola_pav | ok |
| 19:02.36 | nicola_pav | thank u |
| 19:04.10 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
| 19:05.48 | *** part/#asterisk sikanrong (n=sikanron@8.104.219.87.dynamic.jazztel.es) |
| 19:06.16 | *** join/#asterisk ThatKidKel (n=Kelvin@208.110.55.5) |
| 19:07.17 | ThatKidKel | does anyone know if the polycom phones support a central directory that comes from a website? ie a database driven site that outputs xml? instead of the 00000000-directory.xml file for each phone? |
| 19:07.37 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:08.36 | SuPrSluG | they support http provisioning |
| 19:08.59 | ThatKidKel | where can i find a link to that? |
| 19:09.22 | [TK]D-Fender | ThatKidKel: Yes, in 3.X firmwares there is an option for LDAP based corporate directories, but that is a licensed option |
| 19:09.26 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 19:09.33 | [TK]D-Fender | ThatKidKel: Nothing free |
| 19:09.44 | ThatKidKel | ah |
| 19:09.54 | [TK]D-Fender | ThatKidKel: A possibly viable alternative is to use a MicroBrowser script |
| 19:09.57 | ThatKidKel | i was hoping there was something like the 79xx xml services |
| 19:10.14 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
| 19:10.25 | [TK]D-Fender | ThatKidKel: there is, like I jsut said |
| 19:11.12 | *** part/#asterisk lou_gr (n=louben@212-70-216-131.ath.static.tee.gr) |
| 19:11.43 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 19:12.41 | kannan | hello, i am planning to get cepstral license (1 port with voice) for test-to-speech. Will this restrict the port to one call channels at a time , or is it only to "process" the message; Ie: it will take 1 second to synthesize speech taht will be delivered to user in phone call |
| 19:13.46 | *** join/#asterisk Hemos\ (n=cyberspa@82.193.31.157) |
| 19:14.02 | neurosys | [TK]D-Fender: sorry for the delay. Thanks brother! |
| 19:15.41 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:16.49 | neurosys | [TK]D-Fender: The rooms are already wired for analog :( |
| 19:19.51 | *** join/#asterisk SebastianS (n=schu@adsl-dyn215.78-98-80.t-com.sk) |
| 19:20.52 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 19:23.35 | tzafrir_laptop | technically you'll need an extra FXS port for each phone |
| 19:25.01 | bmoraca | neurosys: that's why you use a SIP MEDIA GATEWAY...it converts SIP to FXS |
| 19:26.47 | bmoraca | FSX...otherwise known as the port an analog phone needs to plug in to in order to get dial tone |
| 19:26.59 | bmoraca | blah |
| 19:27.11 | bmoraca | FXS...can't type today |
| 19:29.05 | KyleK | kannan: that might depend on how you use it? they have a demo available so you should be able to see how long it takes to write out a long sentance |
| 19:29.35 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 19:29.45 | tzafrir_laptop | bmoraca, sip to FXS? |
| 19:29.52 | tzafrir_laptop | FXO to sip, maybe |
| 19:30.14 | tzafrir_laptop | oops, sorry, my mistake |
| 19:30.19 | bmoraca | tzafrir_laptop: SIP to FXS for connecting to his 28 stations |
| 19:30.34 | neurosys | bmoraca: Thnx man. I do appreciate all of this. |
| 19:31.05 | bmoraca | it's Friday-eve-eve-eve... :) |
| 19:31.47 | KyleK | oh i just got a small problem, I'm doing Dial(SIP/hardhone&SIP/softphone,20) then on to voicemail, I need to add something so it will keep ringing for 20 seconds, can I Dial a null device of some kind? or will i need to set up a sip client on my server that just ignores calls? |
| 19:31.58 | bmoraca | neurosys: although I do agree with tzafrir_laptop...a PRI or multi-channel SIP account would be sooo much easier to deal with than 8 analog lines... |
| 19:32.23 | neurosys | bmoraca: they seem to be locked into a long comcast contract :( |
| 19:32.38 | bmoraca | KyleK: Play(ringing) maybe? |
| 19:32.40 | telnettech | ok i am needing some help |
| 19:32.58 | telnettech | I am trying to dial out on a PRI and Im getting the following results on the CLI |
| 19:33.02 | telnettech | http://pastebin.com/d6b316646 |
| 19:33.03 | bmoraca | neurosys: not always...many SIP accounts are month-to-month. PRIs generally are long term, yes |
| 19:33.18 | telnettech | can someone tell me what cause 100 means? |
| 19:33.29 | telnettech | is that related to the SIP error codes? |
| 19:34.02 | KyleK | bmoraca: that'd also pick up the line, I'm trying to avoid picking up the line instantly when the SIP phones are unvailable as there is a handset thats hooked up to the incoming phone line |
| 19:34.25 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 19:34.51 | bmoraca | telnettech: your PRI provider doesn't understand the number you dialed out |
| 19:34.51 | Nugget | telnet is eeeeeeevil! |
| 19:35.25 | telnettech | funny that they are the people that gave me that nmber to call.....lol and then they complain i am not allowing access to the service |
| 19:35.36 | jaytee | Cause No. 100 - Invalid information element contents |
| 19:35.53 | bmoraca | telnettech: good reference: http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
| 19:36.11 | telnettech | thanks bmoraca |
| 19:36.25 | jaytee | my favorite cause code is 117 which is "Do Not Call During Nap TIme" |
| 19:36.31 | bmoraca | lol |
| 19:36.35 | *** join/#asterisk trentcreek (n=kvirc@200.94.227.117) |
| 19:37.06 | jaytee | or at least that's the cause I was getting when someone on our Asterisk system would call this one number and the woman answering would put the call on hold and then hangup. |
| 19:37.56 | jaytee | When I finally got ahold of her she insisted I not call during Nap Time |
| 19:37.57 | ruben23 | can i install snmp for my bandwidth monitoring on my asterisk server>..does it wont affect the servers perfomance on call...? |
| 19:38.31 | neurosys | bmoraca: please forgive my newbieness. But when you say SIP account, you talkingabout using an ITSP? |
| 19:40.11 | bmoraca | neurosys: yes. |
| 19:41.30 | bmoraca | anyone else going to Interop this week? |
| 19:41.31 | neurosys | bmoraca: but theyre stuck in a contact with comcast for 8 lines. how can i pitch around that? :P |
| 19:41.43 | neurosys | bmoraca: Where is Interop going to be? |
| 19:43.14 | bmoraca | tell them how much it's going to cost to get an 8 port TDM card with echo cancelling and then tell comcast that you're reducing service from 8 analog lines down to 1 analog line (for fax/emergency) and comcast won't really be able to do anything about it, because you're technically still in contract. |
| 19:45.40 | jplank | am I going crazy? is there no way with web-meetme for the moderator to mute all users on the conference? |
| 19:48.04 | jaytee | whether the moderator can or cannot mute users has no bearing on your sanity |
| 19:50.09 | neurosys | bmoraca: slick. smart move. Thnx |
| 19:50.27 | jplank | tell that to my client |
| 19:56.48 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:07.02 | trentcreek | I was trying ยจmake uninstallยจ" and "sudo make uninstall" for "asterisk-addons-1.4.8 " but it returns "No rule to make target `uinstall' " Any alternatives? |
| 20:10.32 | voxter | anyone here successfully have a cisco 7941/7961 working behind nat to an asterisk server? |
| 20:12.21 | bmoraca | voxter: 7941, no, but Cisco phones are very particular about their NAT settings. you have to tell the phone that it's going to be behind a NAT as well as telling the server that the phone is behind a NAT |
| 20:14.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:16.49 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 20:18.28 | *** join/#asterisk KyleK (n=Kyle@allspark.shadowmage.org) |
| 20:18.47 | Nugget | for any modern cisco phone (79x2 or higher) you need to use sip-tcp to get it to play with nat |
| 20:19.23 | Nugget | 79x1 might be similarly afflicted. |
| 20:19.59 | Nugget | if it's an xml configuration file, you're probably screwed. :) |
| 20:20.28 | *** join/#asterisk jasonpr (n=jasonpr@216.58.247.235) |
| 20:20.33 | jasonpr | Hey I'm trying to decide what to write a new OSS app in. I'd like to take a quick poll. Who here likes Java/PHP |
| 20:20.47 | KyleK | how can I configure the delay between dialing on the handset and call creation with a SIP3102 device? |
| 20:21.40 | KyleK | jasonpr: if its a local app write it in java, if its web based maybe php? |
| 20:21.56 | Corydon76-dig | jasonpr: Write it in C. |
| 20:22.07 | Nugget | Jave > PHP but just barely. they're both pretty far down the list for me. |
| 20:22.10 | Nugget | er, Java |
| 20:22.14 | Nugget | PHP is particularly heinous |
| 20:22.22 | paulius | jasonpr: I like PHP for web. |
| 20:22.28 | KyleK | I love heinous for personal use |
| 20:22.30 | Nugget | and unless you're talking about a web app, PHP is *absolutely* the wrong choice. |
| 20:22.33 | paulius | Scales well, amazing performance, and widespread adoption of PHP. |
| 20:22.38 | voxter | Nugget: yeah i think i might just replace it with a 7940 for this user. the 7941 has 'kind of registered' but calls are screwed up and it constantly times out,e tc |
| 20:23.21 | Nugget | voxter: it's not tough to do sip-tcp, if you're on asterisk 1.6, but downgrading to 79x0s is a perfectly sane approach. that's what I do here |
| 20:23.28 | Nugget | (for NATted phones) |
| 20:23.43 | jasonpr | The app needs to scale and it needs to be easily configurable for a few users |
| 20:23.45 | voxter | Nugget: cant do asterisk 1.6, this client is on a small soekris embedded box which is 1.2 based |
| 20:23.58 | Nugget | *nod* 7940 it is, then. |
| 20:24.34 | paulius | But isn't the screen way better on the 7941? :-P |
| 20:24.42 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 20:25.35 | Nugget | they both suck compared to a 7945. :) |
| 20:25.48 | paulius | pets his 7971 |
| 20:25.50 | paulius | you were saying? |
| 20:27.07 | *** join/#asterisk bullium (n=RAIN\wbr@216.68.250.30) |
| 20:29.57 | bullium | I've got an issue whenever a SIP extension is offline instead of going straight to voicemail it will wait the standard 20secs (silence on the callers end) and then drop the call into voicemail. How can I make it drop the call to voicemail if the SIP user isn't connected dialplan example (exten = 6187,1,Dial(SIP/6187,20,i) |
| 20:29.57 | bullium | exten = 6187,n,Voicemail(6187) |
| 20:32.42 | *** join/#asterisk mythicalbox (n=mitchel@adsl-67-167-10.hsv.bellsouth.net) |
| 20:33.01 | mythicalbox | hello, i'm experiencing one-way audio with asterisk |
| 20:33.42 | mythicalbox | the odd part is that there isn't "really" any NAT involved here.. I have two Cisco ASA devices connected to eachother to form a site to site VPN |
| 20:34.05 | mythicalbox | so both, the phone (a cisco 7940) and Asterisk can see eachother directly |
| 20:34.15 | jasonpr | mythicalbox: SIP? |
| 20:35.10 | jasonpr | mythicalbox: Are you using SIP? |
| 20:35.27 | mythicalbox | yes, SIP |
| 20:35.39 | jasonpr | well if you are you need to look in your sip.conf |
| 20:35.59 | jasonpr | there is a range of UDP ports that need to forwarded from your router |
| 20:36.12 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 20:36.14 | jasonpr | assuming asterisk and your phone are behind the firewall |
| 20:36.22 | jasonpr | you can change the port range if you don't like them |
| 20:36.39 | mythicalbox | there is no firewall |
| 20:36.48 | jameswf | b14ck: STOP IT |
| 20:36.50 | jameswf | :) |
| 20:36.51 | beek | jaytee: ping |
| 20:37.00 | mythicalbox | i just looked at [general] i think it could be "localnet" |
| 20:37.13 | mythicalbox | i'll try setting it to both networks |
| 20:37.29 | jaytee | beek: pong |
| 20:37.41 | beek | jaytee: Mind if I PM you? |
| 20:37.45 | jaytee | sure |
| 20:38.56 | b14ck | jameswf, lol |
| 20:43.26 | telnettech | so the telco tells me that the *510 is not what they want, they want xxx-xxx-xxxx |
| 20:43.46 | telnettech | how the hell am i to know that if you tell me that the guests should be able to dial *510 for the service |
| 20:44.46 | telnettech | that you want xxx-xxx-xxxx unles you tell me!!!!! |
| 20:46.17 | *** join/#asterisk Micho123 (n=mcho123@94.187.118.22) |
| 20:46.35 | *** part/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 20:47.54 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:48.44 | Micho123 | Hi all, I'm using asterisk in realtime and all my dialplans are defined in extensions table...My problem is the following...If I have for a specific context an extension with value _X. and another entry for the same context with extension 123456, if an incoming call from the same context comes to asterisk to extension 123456 the _X. is matched first...is there a way to force asterisk to match the 12345 extension first before _X. match? |
| 20:48.53 | *** join/#asterisk nealix (n=np20433@nat/sun/x-c484fe5d35b22940) |
| 20:49.32 | mythicalbox | jasonpr: adding a network to localnet fixed the issue, thanks for jogging my memory :) |
| 20:50.25 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:50.40 | rhassing | Micho123, If you put it in front of the _X. it should work, or create an other context and put the _X. in there and include it in the first context |
| 20:51.23 | Micho123 | rhassing, No include... |
| 20:51.46 | rhassing | why don't you want the use an include? |
| 20:52.31 | *** join/#asterisk ctp (n=ctp@brsg-d9bee524.pool.mediaWays.net) |
| 20:53.48 | Micho123 | rhassing, do you mean include a context inside another one? |
| 20:54.29 | rhassing | Micho123, yes, just to make sure the context is checked and will jump to the included context next |
| 20:54.55 | Micho123 | rhassing, let me explain what i have |
| 20:56.00 | Micho123 | rhassing, extension 123456 has BE context...this context is used for outgoing calls from this context...the dialplan for that context is : _X. dial(SIP/....) |
| 20:56.13 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 20:57.31 | Micho123 | rhassing, I have another dialplan with context IN...When the call come to IN context when dialing 123456, asterisk match BE context with _X. instead of matching an entry 12345 goto Context |
| 20:59.33 | rhassing | Micho123, To which context does the peer belong where this call comes from? |
| 20:59.50 | Micho123 | rhassing, IN context |
| 21:00.53 | rhassing | Micho123, and if you do a "dialplan show IN" |
| 21:01.04 | *** join/#asterisk b14ck (n=blacky@cpe-98-151-210-28.socal.res.rr.com) |
| 21:01.11 | bmoraca | Micho123: this is why it's generally recommended NOT to use such broad patterns as _X. |
| 21:01.41 | Micho123 | bmoraca, mmmm |
| 21:02.16 | bmoraca | only match the patterns you need to, and match them exactly |
| 21:02.19 | joesuffceren | what zaptel module should I load for a TE210P? |
| 21:02.56 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 21:03.06 | rhassing | Micho123, That's also true :-) Make sure the patterns are as exact as possible |
| 21:03.18 | jaytee | quittin time, be back later |
| 21:04.12 | rhassing | Micho123, But then again, it should not get to this context if the peer and context are correctly configured |
| 21:04.25 | bmoraca | Micho123: think of it like IP routing...would you route an entire /16 when your target is a /24? no...so why would you want to match 123 when your target is 8885551234 |
| 21:05.57 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 21:06.16 | jplank | in a meetme is there any way to dynamically set monitor mode, during the call, or a way to mute everyone but the moderator? |
| 21:06.17 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 21:07.14 | lesouvage | jplank: use the a parameter for the one that has to be the admin and press * as admin. That starts a menu |
| 21:08.53 | jplank | but the * menu only allows the moderator to mute themselves, not everyone else |
| 21:13.52 | *** join/#asterisk Flyser (n=Flyser@unaffiliated/flyser) |
| 21:16.27 | *** join/#asterisk donavan (n=donavan@centos/slackers/donavan) |
| 21:17.16 | donavan | you people build packages that don't have uninstall scripts and then send your users to OS channel for support? |
| 21:17.25 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:18.02 | drmessano | ? |
| 21:18.09 | drmessano | donavan: Say what? |
| 21:18.32 | Chainsaw | drmessano: donavan demands to see the emperor about this lack of an uninstall target. And where's his pony! |
| 21:19.50 | drmessano | Last time I checked, Digium didn't build packages.. except those RPMs in AsteriskNOW |
| 21:20.05 | drmessano | I guess it's 4:20 there |
| 21:20.10 | trentcreek | drmessano: That VPS ran out of memory and acted like windows....no swap...boooo.. |
| 21:20.24 | drmessano | nice |
| 21:20.49 | lesouvage | jplank: " 'N' -- Mute all non-admin users in the conference\n" is in the app_meetme.c . But how to run an admin command? |
| 21:22.16 | donavan | drmessano: who provides the -addons tarball ? |
| 21:22.39 | drmessano | Tarball is not a package.. its a tarball |
| 21:22.41 | [TK]D-Fender | donavan: depends which tarball you're talking about |
| 21:22.55 | [TK]D-Fender | donavan: I could make you a taball right now... |
| 21:23.01 | trentcreek | addons.1.4.8 |
| 21:23.22 | drmessano | Why the hell would you need an UNINSTALL for ADDONS? |
| 21:23.33 | drmessano | Its like a few modules.. done |
| 21:23.36 | *** join/#asterisk vimpulse (n=ask_me@CPE00134609fbfb-CM0013718690da.cpe.net.cable.rogers.com) |
| 21:23.43 | donavan | drmessano: ask your newest user trentcreek |
| 21:24.22 | drmessano | rm ? |
| 21:24.26 | tzafrir_laptop | donavan, missing uninstall target for what exactly? |
| 21:24.52 | seanbright | heh |
| 21:25.14 | jplank | I see that in show application meetmeadmin, but how would I use that during a call? |
| 21:25.36 | drmessano | No wonder the devs carry shotguns with them to cons. |
| 21:25.45 | trentcreek | tzafrir_laptop: addons 1.4.8 |
| 21:25.50 | seanbright | kaii: apologize to donavan for sending trentcreek to #centos |
| 21:25.53 | seanbright | :) |
| 21:25.58 | drmessano | Not like theres real issues in Asterisk to argue about... "Why is the A capitalized" |
| 21:26.08 | seanbright | how do you pronounce dahdi?? |
| 21:26.10 | donavan | tzafrir_laptop: normal source packages (tarballs included) that contain a make install target include a make uninstall target |
| 21:26.27 | drmessano | ahem |
| 21:26.42 | seanbright | that is not true |
| 21:26.49 | seanbright | subversion for example |
| 21:26.50 | tzafrir_laptop | considers 'make uninstall' dodgy at best |
| 21:26.56 | seanbright | source tarball has 'make install' |
| 21:26.56 | jplank | would I have to recompile app_meetme.c with that option in the menu? |
| 21:26.58 | seanbright | no make uninstall |
| 21:27.25 | vimpulse | tzafrir_laptop, why do you consider 'make uninstall' dodgy? |
| 21:28.23 | tzafrir_laptop | in dahdi: make KVERS=2.6.20 install; make KVERS=2.6.21 uninstall |
| 21:28.35 | tzafrir_laptop | or, in a more realistic scenario: |
| 21:29.10 | tzafrir_laptop | you install when your kernel is 2.6.20. you just run 'make install' |
| 21:29.53 | tzafrir_laptop | later on you install 2.6.21 and boot to it. now you run 'make uninstall' . and you have erased the dahdi modules from kernel 2.6.21 |
| 21:30.37 | tzafrir_laptop | The state of the system when you install is not the one when you uninstall |
| 21:30.47 | seanbright | not to mention... i gave trentcreek the necessary commands to uninstall addons 1.4.8 |
| 21:31.10 | seanbright | MYMODDIR=/path/to/modules |
| 21:31.11 | seanbright | for x in res_config_mysql app_addon_sql_mysql app_saycountpl cdr_addon_mysql chan_ooh323 format_mp3; do rm ${MYMODDIR}/${x}.so; done |
| 21:31.14 | seanbright | BAM |
| 21:31.18 | seanbright | (the BAM is optional) |
| 21:32.21 | *** join/#asterisk jtodd (n=jtodd@c-76-124-123-117.hsd1.pa.comcast.net) |
| 21:32.21 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:33.07 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 21:34.07 | tzafrir_laptop | make: BAM: Command not found |
| 21:34.07 | tzafrir_laptop | make: *** [all] Error 127 |
| 21:34.24 | seanbright | that's why it's optional |
| 21:34.28 | KyleK | why does www.dundi.com just redirect to digium.com now? was dundi given up on? |
| 21:34.36 | Nugget | you need to apt-get emeril if you want the BAM command. |
| 21:34.51 | donavan | tzafrir_laptop: a kernel removal (using sane package management) will not fiddle with files it does not own |
| 21:35.19 | vimpulse | tzafrir_laptop: fair point. OK, I don't know about dahdi, but for non-kernel-related software, dpkg deals with that by always uninstalling the old version before installing the new version. If I observe the same safeguard, then would "make uninstall" be reliable for me? |
| 21:35.20 | tzafrir_laptop | donavan, it's not the kernel romoval . it's the uninstall target of dahdi |
| 21:35.59 | lesouvage | jplank: there is not much documentatio to be found. There is a whole list of parameters in the code for admin commands and i guess you can use them with the asterisk management interface (AMI) by webpage for example but i'm not sure about that. |
| 21:36.03 | tzafrir_laptop | vimpulse, dpkg, rpm and other have their own bookkeeping . This is not a simple 'make uninstall' |
| 21:36.33 | donavan | tzafrir_laptop: source installs never touch rpm or dpkg databases |
| 21:36.48 | paulius | Can someone ping the bot to fetch a link of trunks? |
| 21:36.51 | paulius | *SIP trunks that is |
| 21:36.53 | jtodd | KyleK: I believe that the dundi.com pages are incorrectly redirected at the moment; there is an inquiry to the people responsible for DNS and virtual hosting here at Digium. |
| 21:36.53 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 21:37.08 | tzafrir_laptop | donavan, in that case, change the default of asterisk to install to /usr/local rather than /usr |
| 21:37.18 | vimpulse | tzafrir_laptop: could people who write "make uninstall" functions easily design in the same bookkeeping? |
| 21:37.27 | donavan | and in the scenarios of clueless user who issue make install as root, should have a make uninstall target available to clean up their abortion |
| 21:37.40 | *** join/#asterisk docid (n=frankie@whthyt253-26.northwestel.net) |
| 21:37.40 | tzafrir_laptop | vimpulse, there are some 'make install' wrappers to do that |
| 21:37.57 | tzafrir_laptop | e.g. stow, IIRC |
| 21:38.09 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 21:38.53 | *** part/#asterisk jtodd (n=jtodd@c-76-124-123-117.hsd1.pa.comcast.net) |
| 21:39.07 | docid | hrmm, so im trying to identify how much real wold bandwith a call over iax2 in gsm format takes up , and if it combines headers etc, ie less extra bandwith for the second third etc call, to do this i need to isolate the network between the 2 asterisk boxes, but if i do that asterisk cannot talk to the atas or ip phones, so im wondering how to simulate multiple calls from the cli |
| 21:39.08 | vimpulse | tzafrir_laptop: true. So I guess it is not worthwhile for package maintainers to do the same bookkeeping that stow does. :) |
| 21:39.33 | CGMChris | I have been working with my SIP provider regarding T.38 and I keep running into problems. They complain that Asterisk is choosing an odd numbered RTP port or sending only the port number (and not the IP address) in the 200 OK response. Is anyone familiar with T.38 issues or perhaps there is documentation that isnt indexed by Google regarding this? |
| 21:39.37 | tzafrir_laptop | vimpulse, what do you mean by package maintainers? asterisk or dpkg/rpm ? |
| 21:39.47 | seanbright | people need to stop saying 'package' when they mean tarball |
| 21:40.01 | donavan | going to peacefully depart |
| 21:40.15 | jplank | lesouvage: I'm having the same trouble. I get how to do it through the AMI, and really, web-meetme3 already provides that functionality, but for the life of me I can't figure out how to do it through the phone. Maybe I have to patch app_meetme.c to add that as an option to the admin menu, and then recompile? |
| 21:41.10 | vimpulse | tzafrir_laptop: oops, I should have said "upstream developers" instead of "package maintainers". |
| 21:42.06 | *** join/#asterisk jtodd (n=Adium@c-76-124-123-117.hsd1.pa.comcast.net) |
| 21:42.06 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 21:43.24 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 21:43.43 | docid | so im trying to identify how much real wold bandwith a call over iax2 in gsm format takes up , and if it combines headers etc, ie less extra bandwith for the second third etc call, to do this i need to isolate the network between the 2 asterisk boxes, but if i do that asterisk cannot talk to the atas or ip phones, so im wondering how to simulate multiple calls from the cli |
| 21:43.51 | kaii | donavan: i'm sorry for sending ${namehere} to #centos |
| 21:43.56 | seanbright | heh |
| 21:44.09 | CGMChris | seanbright: know anyone that could answer a few T.38 questions? |
| 21:44.09 | kaii | :) |
| 21:44.18 | seanbright | CGMChris: not me |
| 21:44.25 | drmessano | kaii: Find a better "blowoff" channel.. Sometimes, they throw the fish back and chase it |
| 21:44.38 | seanbright | CGMChris: working on an uninstall target for asterisk addons |
| 21:44.43 | seanbright | (but not really. just trolling.) |
| 21:44.45 | drmessano | kaii: #trixbox is kinda demeaning.. Keep it handy |
| 21:44.51 | kaii | lol |
| 21:44.56 | donavan | ##linux is handy |
| 21:45.04 | [TK]D-Fender | docid: A single call over GSM is no different than over SIP/RTP |
| 21:45.04 | lesouvage | docid: see http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 21:45.18 | drmessano | ##linux works too.. So does #newb |
| 21:45.31 | seanbright | #2,000 |
| 21:45.33 | CGMChris | [TK]D-Fender: Are you familiar enough with T.38 to answer a question or 2? |
| 21:45.53 | trentcreek | Now I see what is going on....My modules are not loading...who can point me in the right direction? |
| 21:45.54 | [TK]D-Fender | docid: And seriously... don't repeat the same question every 5 minutes... |
| 21:46.04 | [TK]D-Fender | CGMChris: not much |
| 21:46.11 | seanbright | trentcreek: hey buddy! i can! |
| 21:46.13 | seanbright | heh |
| 21:46.22 | trentcreek | groovy |
| 21:46.22 | KyleK | seanbright: does #2,000 exist on freenode? |
| 21:46.27 | seanbright | KyleK: sure does |
| 21:46.33 | drmessano | seanbright: Back off, I think he needs a developer to help |
| 21:46.37 | *** part/#asterisk vimpulse (n=ask_me@CPE00134609fbfb-CM0013718690da.cpe.net.cable.rogers.com) |
| 21:46.38 | KyleK | sometimes the IRCd blocks such a cool channel :) |
| 21:46.44 | seanbright | drmessano: tip of the hat, sir. |
| 21:46.45 | docid | hrmm, so ive tried to tell the boss, but it seems that he wants real numbers, sorry for the repeat, i saw a couple of people had joined the channel and i had no response previously, so it was more posing it to the new people |
| 21:46.53 | CGMChris | seanbright: perhaps you have a developer who has worked with T.38, or someone who wrote docs, or something. Surely someone knows about T.38? |
| 21:47.06 | *** join/#asterisk Docteh (n=mage@24.86.183.239) |
| 21:47.07 | drmessano | My fav IRC channel is #0 |
| 21:47.15 | seanbright | CGMChris: i know approximately squat about all things |
| 21:47.18 | Qwell | donavan: I'm not sure I see the problem... |
| 21:47.31 | donavan | Qwell: where? |
| 21:47.32 | CGMChris | seanbright: I thought you were the guy who coordinates the devs? |
| 21:47.56 | seanbright | Qwell: while he's here, ask him about the whole 32 bit packages satisfying 64 bit depends! yay! |
| 21:47.59 | Docteh | seanbright: doesn't work as intended |
| 21:48.06 | Qwell | donavan: normally you'd flog a user and be done. we've both told him he isn't running CentOS on several occasions... |
| 21:48.08 | seanbright | CGMChris: good god, no. |
| 21:48.47 | seanbright | trentcreek: did you update astmoddir in /etc/asterisk/asterisk.conf |
| 21:48.48 | trentcreek | module show returns '--- 0 modules loaded |
| 21:48.49 | donavan | Qwell: the lack of uninstall targets irritated me into action |
| 21:48.51 | seanbright | Docteh: eh? |
| 21:48.56 | trentcreek | seanbright: yes I did |
| 21:49.04 | [TK]D-Fender | docid: New people? You mean all of... jtodd? |
| 21:49.08 | [TK]D-Fender | docid: DON'T |
| 21:49.09 | Docteh | <PROTECTED> |
| 21:49.18 | [TK]D-Fender | docid: And as I said, same math as for RTP. |
| 21:49.24 | *** join/#asterisk ctp (n=ctp@brsg-d9bee524.pool.mediaWays.net) |
| 21:49.30 | seanbright | Docteh: ah. yes. that was intentional. |
| 21:49.39 | Qwell | donavan: It's like asking for an uninstall target for half of KDE. |
| 21:49.40 | [TK]D-Fender | docid: 13+21 = 33kbps |
| 21:50.01 | Qwell | donavan: it just doesn't make sense in this instance. Asterisk proper does have an uninstall target |
| 21:50.03 | Docteh | seanbright: on some ircd's the net result is a /partall |
| 21:50.15 | seanbright | trentcreek: run asterisk from the command line (sudo asterisk -cvvvvvvvvvvvvvg) and pastebin *all* of the output |
| 21:50.18 | docid | ok, thanks much for the answer, |
| 21:50.18 | seanbright | ~pb |
| 21:50.19 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 21:50.40 | [TK]D-Fender | docid: 33/34 |
| 21:50.57 | docid | so umm, does that meen ther isnt a way you know of to simulate calls from the command line? boss is asking for real numbers on our install....doesnt trust calcs.... |
| 21:50.59 | [TK]D-Fender | docid: This is all on the WIKI page for codec bandwidth consumption |
| 21:51.05 | Qwell | donavan: imagine how absurd it would be to say "make an uninstall target for KDE games". make sense? |
| 21:51.09 | docid | ahh, must have misssed that |
| 21:51.21 | Qwell | a distro could certainly package it so that's possible, but... |
| 21:51.30 | donavan | Qwell: is there a make install target in that source package? |
| 21:51.36 | [TK]D-Fender | docid: Real numbers? You boss is a MORON. Bandwidth is bandwidth. |
| 21:51.47 | Qwell | Yes there is. |
| 21:51.49 | docid | well, im not denying that |
| 21:51.53 | trentcreek | seanbright: http://pastebin.ca/1427877 |
| 21:51.55 | seanbright | Qwell: it's a trap!! |
| 21:51.56 | [TK]D-Fender | docid: This isn't like benckmarking a hard-drive |
| 21:52.06 | donavan | Qwell: then there should be a Make uninstall |
| 21:52.09 | [TK]D-Fender | docid: There is no "real world" vs "theory" |
| 21:52.15 | [TK]D-Fender | docid: it is FIXED. |
| 21:52.18 | docid | so ive tryed to tell him |
| 21:52.27 | seanbright | trentcreek: pastebin your /etc/asterisk/modules.conf file |
| 21:52.45 | [TK]D-Fender | docid: How fast is 45 RPM in the real world? |
| 21:52.48 | docid | once again, thanks, ill try and explain things yet again |
| 21:52.54 | Qwell | donavan: patches welcome |
| 21:52.54 | ajohnson | [TK]D-Fender: Yeah, but due to kerning, 0's are bigger than 1's... so GSM could have different amounts of bandwidth than "theoretical" tests... |
| 21:52.55 | lesouvage | docid: are you aware of the fact that listings of this irc conversation are on the internet. |
| 21:53.01 | ajohnson | ;-) |
| 21:53.08 | docid | depends, if its quotes as bare motor speed on a blender, its more like 33-25 |
| 21:53.17 | trentcreek | seanbright: http://pastebin.ca/1427880 |
| 21:53.18 | donavan | Qwell: I know how to use /ban |
| 21:53.26 | docid | but i get your point |
| 21:53.27 | [TK]D-Fender | ajohnson: 0's take up more in on a printed page and more with, so 0 is bigger than 1, right? |
| 21:53.27 | lucasb | Good afternoon! Does anyone know of a good website that describes how to set up Asterisk as a PRI gateway? I have a Sangoma A102D T1/PRI card. |
| 21:53.35 | [TK]D-Fender | ink* |
| 21:53.36 | Qwell | donavan: Are you seriously threatening to ban me from #centos because I disagree with your point in #asterisk? |
| 21:53.39 | ajohnson | yes! |
| 21:53.58 | donavan | negative Qwell |
| 21:54.14 | Qwell | Then by all means, elaborate. |
| 21:54.19 | lesouvage | lucasb: afaik sangoma has petty good istruction for doing this on its website. |
| 21:54.33 | donavan | to +b the noise in #centos from asterisk victims |
| 21:54.35 | lucasb | lesouvage, thanks, will check that out |
| 21:54.35 | trentcreek | seanbright: sorry..i did not paste it all..hand on |
| 21:54.41 | rhassing | lucasb, www.voip-info.org |
| 21:54.48 | [TK]D-Fender | lucasb: Sangoma's wiki describes how to config their cards for use with Zaptel/DAHDI, the rest is all Asterisk |
| 21:54.58 | drmessano | asterisk "victims".. Ahhh IRC drama |
| 21:55.00 | [TK]D-Fender | lucasb: and the word "gateway" is dangerously vague |
| 21:55.00 | seanbright | victims. nice. |
| 21:55.05 | Qwell | donavan: Nobody here should be sending users there for non-CentOS issues. Please feel free to remove users who aren't following your channels rules. |
| 21:55.06 | lucasb | I know how to configure the card... easy enough.. I'm just new to Asterisk |
| 21:55.09 | drmessano | Thats what this is about.. douchebaggery |
| 21:55.17 | [TK]D-Fender | lucasb: ... |
| 21:55.19 | [TK]D-Fender | ~book |
| 21:55.19 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:55.21 | [TK]D-Fender | lucasb: ^^^ |
| 21:55.29 | *** join/#asterisk M1s3ry (n=jbigbee@boromir.api-digital.com) |
| 21:55.36 | lucasb | [TK]D-Fender, a media gateway... just interfacing the PRI to VoIP.... I have always been using Freeswitch |
| 21:55.53 | bmoraca | wow |
| 21:55.54 | donavan | s/victim/idiots who install and compile source packages without uninstall targets/ |
| 21:56.20 | M1s3ry | howdy, anyone know how to force install a package via yum? IE: yum install kernel-devel-'2.6.18-128.1.1.el5' Trying to get the correct kernel source so that I can make all && make install dahdi... |
| 21:56.24 | seanbright | wow |
| 21:56.25 | Qwell | donavan: You and I both know they shouldn't be installing from source on a packaged distro. |
| 21:56.27 | donavan | thats is a nifty bot |
| 21:56.33 | donavan | s/victim/idiots who install and compile source packages without uninstall targets/ |
| 21:56.37 | drmessano | fascinating |
| 21:56.39 | Qwell | donavan: I see no issue with what you're suggesting. |
| 21:56.42 | seanbright | and to think i thought centos was a turd *before* |
| 21:56.56 | KyleK | hrm |
| 21:57.05 | trentcreek | seanbright: |
| 21:57.08 | trentcreek | http://pastebin.ca/1427884 |
| 21:57.09 | seanbright | trentcreek: |
| 21:57.10 | lesouvage | lucasb: afaik sangoma has petty good instruction for setting up their cards and get the cards up and running. As [TK]D-Fender tells you the rest is indeed configuring Asterisk |
| 21:57.23 | lucasb | k. thanks |
| 21:57.32 | KyleK | s/h/g |
| 21:57.42 | *** join/#asterisk DarkRift (n=dark@65.92.167.128) |
| 21:57.43 | donavan | Qwell: I agree, but what do you do when it happens? |
| 21:57.55 | [TK]D-Fender | lucasb: * isn't jsut a "gateway" that implies a single-purpose transform. |
| 21:58.04 | [TK]D-Fender | lucasb: Although that can be an application. |
| 21:58.11 | Qwell | I can't say I've ever heard somebody ask to remove just asterisk-addons. If I saw it, I would have told them. |
| 21:58.12 | KyleK | whooops i cant do substituion correctly |
| 21:58.13 | [TK]D-Fender | lucasb: Thats what you need to distinguish |
| 21:58.23 | KyleK | s/cant/can/ |
| 21:58.29 | donavan | Qwell: ... |
| 21:58.33 | KyleK | I love that feature |
| 21:58.35 | Qwell | donavan: DAHDI and Asterisk both have uninstall targets. |
| 21:58.41 | lucasb | [TK]D-Fender, Agreed... this is a very simple application that I would be using this for... the rest (signalling, etc) is done via a session border controller |
| 21:58.48 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 21:58.55 | donavan | s/foobar/foo |
| 21:58.58 | donavan | s/foobar/foo/ |
| 21:59.00 | M1s3ry | :/ anybady? |
| 21:59.08 | donavan | novel |
| 21:59.13 | M1s3ry | s/anybady/anybody |
| 21:59.15 | seanbright | trentcreek: try to start asterisk again (this time with 'sudo asterisk -cvvvvvvvvvdddddddg') |
| 21:59.36 | seanbright | needs backreference support |
| 21:59.48 | KyleK | M1s3ry: you might want to just find the rpm and rpm -ipv or w/e |
| 21:59.51 | seanbright | s/(s)u/u\1/ |
| 21:59.55 | seanbright | fail. |
| 22:00.39 | KyleK | M1s3ry: also why wont yum install kernel-devel install the right version? are you using an older kernel? |
| 22:00.41 | trentcreek | seanbright: http://pastebin.ca/1427889 |
| 22:01.05 | seanbright | trentcreek: pastebin asterisk.conf |
| 22:01.10 | seanbright | we're giving pastebin a workout. |
| 22:01.17 | trentcreek | quite so |
| 22:01.47 | lucasb | [TK]D-Fender, The way that I want to configure * will be such that it is literally a gateway to the PRI... only 1 SIP client at this point and that's the remote SBC.... we are looking to install PRI gateways at all locations that don't have IP origination. We have to interface the PSTN for orig. and potentially local termination |
| 22:02.50 | trentcreek | seanbright: http://pastebin.ca/1427893 |
| 22:03.08 | lucasb | [TK]D-Fender, we are currently doing with with Freeswitch, but I wanted to try * to see if there is any difference... just need to wrap my head around * configuration files now |
| 22:03.16 | seanbright | trentcreek: delete "/chan_sip.so" from the line starting with astmoddir |
| 22:03.28 | seanbright | fights off tears of rage |
| 22:03.39 | M1s3ry | @kylek: yes |
| 22:03.39 | [TK]D-Fender | lucasb: I doubt * will be much different in a raw transform capacity.... |
| 22:03.42 | M1s3ry | old kernel \ |
| 22:03.51 | M1s3ry | I just found a nifty command :p |
| 22:03.52 | M1s3ry | remove |
| 22:04.00 | M1s3ry | removed it, and install the specifif |
| 22:04.00 | [TK]D-Fender | lucasb: But you're welcome to go for it |
| 22:04.01 | M1s3ry | worked |
| 22:04.17 | lucasb | [TK]D-Fender, I like to learn new things :) |
| 22:04.26 | M1s3ry | thanks anyways, and I know, the "enter key is not a puncuation mark", sorry |
| 22:05.09 | seanbright | trentcreek: so change: |
| 22:05.09 | seanbright | astmoddir => /usr/lib64/asterisk/modules/chan_sip.so |
| 22:05.09 | seanbright | to: |
| 22:05.09 | seanbright | astmoddir => /usr/lib64/asterisk |
| 22:05.57 | seanbright | errr |
| 22:05.57 | seanbright | astmoddir => /usr/lib64/asterisk/modules |
| 22:05.57 | trentcreek | got it |
| 22:05.57 | seanbright | splendid. |
| 22:05.57 | seanbright | but it still doesn't work? |
| 22:06.14 | trentcreek | seanbright: correct |
| 22:06.24 | seanbright | trentcreek: is the console output any different at all? |
| 22:07.30 | trentcreek | seanbright: a lot more |
| 22:07.38 | seanbright | super. pastebin, por favor. |
| 22:07.53 | seanbright | por favor means "if you don't mind" in german |
| 22:08.05 | seanbright | i'm very worldly. |
| 22:08.17 | trentcreek | seanbright: more like "please" in Spanish |
| 22:08.30 | seanbright | ... |
| 22:08.33 | trentcreek | i need to redirect the output..my buffer not big enough |
| 22:08.55 | bmoraca | sarcasm + internets = fail on many levels |
| 22:09.04 | trentcreek | yes yes |
| 22:09.05 | seanbright | bmoraca: clearly |
| 22:09.16 | seanbright | bmoraca: personally i thought it was comedy gold. |
| 22:09.21 | trentcreek | what is that recording thing? |
| 22:09.26 | bmoraca | lol |
| 22:09.32 | [TK]D-Fender | lucasb: more power to you. * configs a bit less crpytic looking, though there are quite a few of them |
| 22:09.36 | trentcreek | the record the input ouput |
| 22:09.44 | [TK]D-Fender | is off to martial arts |
| 22:09.45 | trentcreek | script.. |
| 22:09.45 | [TK]D-Fender | BBIAB |
| 22:10.16 | *** part/#asterisk donavan (n=donavan@centos/slackers/donavan) |
| 22:10.31 | seanbright | trentcreek: redirect? |
| 22:10.48 | seanbright | or screen? |
| 22:10.52 | bmoraca | linux_command > file.thingymajig |
| 22:10.54 | seanbright | i have no idea what you are asking |
| 22:11.11 | bmoraca | or turn logging on in putty |
| 22:11.29 | seanbright | or download paste2pastbin.pl |
| 22:11.32 | seanbright | and just pipe the mother |
| 22:11.40 | seanbright | paste2pastebin.pl, rather. |
| 22:12.42 | KyleK | does voicemail_imap work with gmail? |
| 22:12.54 | seanbright | i'm also willing to be given root access to your machine |
| 22:12.59 | seanbright | because i'm a generous kinda guy |
| 22:13.14 | trentcreek | seanbright: http://pastebin.ca/1427908 |
| 22:13.16 | trentcreek | too late |
| 22:13.38 | trentcreek | i used script |
| 22:14.36 | seanbright | ok... |
| 22:14.45 | seanbright | so what exactly doesn't work? it looks fine |
| 22:16.04 | trentcreek | seanbright: let me see again |
| 22:17.00 | trentcreek | seanbright: okay..it is functioning...thanks so much for the assist |
| 22:17.04 | seanbright | my pleasure |
| 22:17.20 | trentcreek | I ad been fighting this thing for 5 days |
| 22:18.06 | seanbright | damn |
| 22:18.13 | seanbright | well. glad i could help. |
| 22:18.46 | bmoraca | el oh el |
| 22:18.50 | Qwell | trentcreek: What distro is this? |
| 22:19.04 | seanbright | FrankenPBX? |
| 22:19.08 | seanbright | that's what he said in #-dev |
| 22:19.15 | seanbright | no clue what that is. |
| 22:19.15 | trentcreek | Not anymore.. |
| 22:19.28 | trentcreek | It is just straight CentOS now |
| 22:19.33 | Qwell | trentcreek: The #centos people are right. Stop installing software from source and unknown locations. |
| 22:19.43 | Qwell | It's going to continue breaking your systems. Just don't. |
| 22:19.56 | seanbright | there is a newer version of centos out now which is soooo much better |
| 22:19.56 | trentcreek | But I had to use Binaries to install asterisk |
| 22:20.02 | seanbright | they code named it 'ubuntu' |
| 22:20.03 | Qwell | What binaries? |
| 22:20.04 | trentcreek | I have 5.3 |
| 22:20.06 | seanbright | download today |
| 22:20.07 | seanbright | heh |
| 22:20.15 | KyleK | ugh 5.3 is out? |
| 22:20.16 | Qwell | How did you install Asterisk? |
| 22:20.27 | bmoraca | centos > ubuntu for servers...you can't argue that |
| 22:20.33 | trentcreek | From atrpms.net/ |
| 22:20.36 | Qwell | bmoraca: Yes I can. |
| 22:20.40 | KyleK | I help some people out with servers and they use centos |
| 22:20.50 | Qwell | trentcreek: Then you need to install the other asterisk-related packages from there. |
| 22:20.59 | KyleK | i have to look up every little thing since im a debian/ubuntu guy :( |
| 22:21.08 | seanbright | centos is a turd. |
| 22:21.16 | seanbright | that's a technical term. |
| 22:21.18 | seanbright | turd. |
| 22:21.20 | trentcreek | yes..I did ยจ"yum install asterisk*" |
| 22:21.35 | Chainsaw | seanbright: Technically Uninteresting Rip-off Distribution? |
| 22:21.38 | Qwell | You don't understand how the packages are built, and you don't understand how to fix small problems with them... This is why third-party repos and building from source is not recommended. |
| 22:22.03 | trentcreek | Qwell: because they will not compile on this VPS |
| 22:22.10 | Qwell | Then don't run a VPS. |
| 22:22.12 | trentcreek | it is working fine now... |
| 22:22.15 | trentcreek | I have to |
| 22:22.15 | Qwell | You've also been told that. |
| 22:22.23 | trentcreek | Just a poor student |
| 22:22.24 | Qwell | That isn't CentOS. |
| 22:22.25 | Deeewayne | ~nowwhat |
| 22:22.26 | infobot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=FJ3oHpup-pk |
| 22:22.52 | trentcreek | yes..."basterized version" |
| 22:22.52 | Qwell | trentcreek: If you need support in the future - I highly recommend talking to your VPS provider. They are making you jump through these hoops. |
| 22:23.01 | *** join/#asterisk drudge` (i=anonymou@unaffiliated/drudge/x-837452) |
| 22:23.10 | Qwell | They can fix them. I'm going to refuse to, as #centos already has. |
| 22:23.22 | drudge` | how many people have asked about rotary phones and using an IVR? |
| 22:23.30 | drmessano | 0 |
| 22:23.34 | Qwell | drudge`: far more than you'd guess |
| 22:23.34 | seanbright | 1 |
| 22:23.39 | seanbright | at least 1 |
| 22:23.43 | seanbright | counting drudge` |
| 22:23.43 | drudge` | this dude keeps pestering me about it |
| 22:23.48 | drmessano | 0.9 |
| 22:24.01 | seanbright | as drmessano approaches infinite |
| 22:24.03 | Qwell | drudge`: okay, and? Asterisk supports it. What's your question? |
| 22:24.49 | drudge` | i didnt see any quick and easy way through PBX to send the caller to a ring group or queue if there is no response at an ivr menu |
| 22:25.04 | Deeewayne | hated dialing 9's on rotary phones |
| 22:25.10 | Qwell | drudge`: like the 't' extension? |
| 22:25.27 | Qwell | Deeewayne: I lived in the 909 areacode. That's the hardest (besides 900) to dial. |
| 22:25.29 | drudge` | no idea, qwell |
| 22:25.34 | Qwell | ~book |
| 22:25.35 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:25.41 | Qwell | drudge`: It's all in there |
| 22:26.27 | Deeewayne | Qwell, I was in 201 during my rotary days |
| 22:26.55 | trentcreek | Qwell: yes..they keep telling me to find my own template to provide..they have yet to respond to my latest request. It is cheaper to run a $15 a month VPS than to have a box in my apartment running 24/7 using electricty, plus the A/C removing the heat |
| 22:27.05 | drudge` | i got that asterisk book ffrom oriely |
| 22:27.12 | drudge` | i just need to look for 't' extension? |
| 22:27.15 | Qwell | trentcreek: That isn't our problem. Take it up with them. |
| 22:27.25 | trentcreek | yes I know...yes telling you my story |
| 22:27.27 | Qwell | drudge`: That's one way to do that, yes. |
| 22:27.46 | Qwell | trentcreek: I know your story. I've heard it many times, because you've required help to fix something you broke over and over. |
| 22:27.57 | drudge` | whats it short for |
| 22:28.02 | Qwell | drudge`: timeout |
| 22:28.08 | drudge` | ohh |
| 22:28.11 | drudge` | ;D |
| 22:28.28 | drmessano | Its not cheaper to run a $15 VPS than have a box |
| 22:28.41 | seanbright | in soviet russia it is |
| 22:28.44 | drmessano | Power is about $5 a month, A/C offset is about half that |
| 22:28.46 | trentcreek | at least for not the same thing |
| 22:28.50 | seanbright | in soviet russia, server boots you |
| 22:28.59 | trentcreek | yes its...you have not seen my electric bill |
| 22:29.06 | seanbright | trentcreek: pastebin it |
| 22:29.16 | drmessano | trentcreek: Wanna see MY FUCKING electric bill for the 10 machines I have here???? |
| 22:29.17 | trentcreek | I dont have a scanner |
| 22:29.18 | drmessano | DO YOU ???? |
| 22:29.19 | seanbright | heh |
| 22:29.22 | trentcreek | no way |
| 22:29.35 | bmoraca | why would you want to run an asterisk box in a VPS as a poor student? |
| 22:29.38 | seanbright | 4 question marks means he is SERIOUS |
| 22:29.50 | trentcreek | just running 1 box part time is costing me almost $100 a month |
| 22:29.58 | drmessano | seanbright: Serious about how trentcreek has ALLL the answers |
| 22:30.08 | drmessano | trentcreek: Bullshit |
| 22:30.14 | seanbright | drmessano: clearly. |
| 22:30.20 | seanbright | same team, same team. |
| 22:30.21 | seanbright | heh |
| 22:30.47 | florz | trentcreek: oh, a 1 kW computer? |
| 22:30.49 | drmessano | $100 to run a box? Clearly you have no clue what your actual energy usage is.. |
| 22:30.54 | florz | trentcreek: what kind of machine is it? |
| 22:30.57 | trentcreek | I am paying .25 cents a Kilowatt hour for electic..and everything in the unit is electric and i live in a hot ass place |
| 22:31.02 | Qwell | s/what.*/ |
| 22:31.06 | trentcreek | i mean total elctric bill |
| 22:31.25 | paulius | You can get a VPS for cheaper than that. |
| 22:31.28 | drmessano | WOW 1/4 of a cent.. Thats like 1/28th what I pay |
| 22:31.40 | bmoraca | our colo charges us $150/mo per cabinet for power :P |
| 22:31.41 | paulius | $8 per month can get you enough for asterisk |
| 22:32.07 | drmessano | Running a box at his place and not BSing about $100 to run it would work too |
| 22:32.09 | trentcreek | so you are saying you are paying $50 kilowatt hour? |
| 22:32.22 | drmessano | trentcreek: You fail at math |
| 22:32.46 | trentcreek | well I got a C in caculus |
| 22:32.52 | drmessano | and speeling |
| 22:33.03 | trentcreek | crappy KB |
| 22:33.03 | seanbright | there are 2 Cs in calculus |
| 22:33.17 | drmessano | seanbright: and there's a 1kw power supply in his PBX |
| 22:33.51 | seanbright | that's not uncommon |
| 22:33.55 | seanbright | ... in soviet russia |
| 22:33.59 | trentcreek | true, ture |
| 22:33.59 | florz | *g* |
| 22:34.01 | drmessano | Im referring to the AC rating |
| 22:34.14 | trentcreek | it's the Tesla model |
| 22:34.20 | seanbright | heh |
| 22:34.30 | seanbright | first funny thing you've said all day |
| 22:34.32 | drmessano | $100... to run a single box |
| 22:34.33 | seanbright | and on that note... |
| 22:34.36 | seanbright | runs off |
| 22:34.41 | Qwell | so trentcreek - next time you have a problem, what are you going to do first? |
| 22:34.49 | trentcreek | no..I mean $100 a month in electric |
| 22:34.51 | seanbright | <PROTECTED> |
| 22:34.54 | drudge` | here in AZ ac is crucial |
| 22:35.01 | trentcreek | Qwell: Google it |
| 22:35.01 | drmessano | Qwell: Unplug his computer to cool the room down by 37 degrees? |
| 22:35.05 | Qwell | trentcreek: then? |
| 22:35.05 | paulius | Because there's a 1kw power supply doesn't mean that it's taking up that much power. |
| 22:35.14 | trentcreek | Qwell: or ask you ;'= |
| 22:35.14 | drudge` | shit gets hot, its liek 112f right now outside |
| 22:35.27 | Qwell | let me msg you seanbright's phone number - one sec |
| 22:35.35 | trentcreek | groovy |
| 22:35.39 | paulius | who's he? |
| 22:35.46 | drmessano | drudge`: Most people need alternating current.. Getting DC from the power company is a bitch |
| 22:35.55 | Qwell | he's the guy who's afraid that I actually might have his number... |
| 22:36.00 | seanbright | you probably do |
| 22:36.03 | seanbright | i know file and russellb do |
| 22:36.07 | seanbright | or did |
| 22:36.15 | Qwell | I don't. I'm not cool enough. :( |
| 22:36.17 | drmessano | I have russellb's home phone number |
| 22:36.23 | drmessano | Not gonna say where I got it |
| 22:36.24 | drudge` | not me |
| 22:36.33 | seanbright | Qwell: or you didn't come to baltimore last year... one of the two |
| 22:36.42 | Qwell | seanbright: could be that |
| 22:36.48 | trentcreek | That could be a good idea....DC power adatper plug on comuter power supplies...bypass the recifier so you can get C directly from say...solar or wind |
| 22:36.50 | paulius | Hey guys, is there any way to make an extension only able to dial through a certain trunk? |
| 22:37.03 | drmessano | yep |
| 22:37.10 | seanbright | i agree with drmessano |
| 22:37.10 | drudge` | yes |
| 22:37.12 | paulius | How? |
| 22:37.16 | seanbright | oh |
| 22:37.20 | drudge` | use a custom context to only go out through whcihever trunk |
| 22:37.21 | Qwell | by telling it to only dial through that one. |
| 22:37.21 | seanbright | now you want to get all complicated. |
| 22:37.22 | file | hey look! I do have seanbright's phone number |
| 22:37.28 | file | seanbright: why do I have this? |
| 22:37.35 | seanbright | file: lost to history |
| 22:37.38 | Qwell | \msg file quick, what is it?! |
| 22:37.40 | drmessano | Oh he's asked in FreePBX |
| 22:37.41 | Qwell | err, mt |
| 22:37.58 | paulius | drmessano: Well I'm running FreePBX, had a plan B if I got flamed in here. |
| 22:38.03 | trentcreek | Great...I need a custom PBX with lots of commenting...say..bu 9 AM tomorrow? |
| 22:38.09 | drudge` | paul, hit me up in msg and ill help ya |
| 22:38.10 | Qwell | paulius: #freepbx |
| 22:38.15 | drmessano | paulius: Not like the question is even relevent here |
| 22:38.18 | seanbright | trentcreek: i'll do it for $10,000 |
| 22:38.34 | Qwell | paulius: and asking in multiple places is considered to be extremely rude. |
| 22:38.38 | seanbright | USD... not sheckles |
| 22:38.49 | trentcreek | what? this is not chan #c? |
| 22:39.39 | paulius | :-( |
| 22:40.28 | trentcreek | seanbright: How about a 10,000,000 zimbawae bill_ |
| 22:40.31 | trentcreek | ? |
| 22:40.38 | Qwell | seanbright: take it - they revaluated. |
| 22:40.46 | seanbright | pass |
| 22:41.03 | seanbright | <PROTECTED> |
| 22:41.05 | drmessano | Qwell: Worth LESS now? |
| 22:41.09 | Qwell | drmessano: other way |
| 22:41.15 | drmessano | wow |
| 22:41.29 | seanbright | file: i have your number too, though. so don't be a hero. |
| 22:41.30 | seanbright | :-P |
| 22:41.50 | drmessano | "Im your huckleberry" |
| 22:41.59 | *** join/#asterisk Tene (n=tene@poipu/supporter/slacker/tene) |
| 22:42.10 | seanbright | leaves for serious |
| 22:42.37 | joesuffceren | just upgraded asterisk to 1.4.24.1. I got a message complaining about modules not installed by this version of asterisk, and I recognize most of them, but I'm not familiar with app_saycountpl.so. any ideas? |
| 22:42.45 | Qwell | joesuffceren: asterisk-addons |
| 22:43.02 | KyleK | hrm this itsp doesn't have easy info on handling incoming calls with asterisk |
| 22:43.19 | joesuffceren | my googling returned something about saying polish words. If no one at my office speaks polish, do I need this? :-) |
| 22:43.24 | Tene | I'm trying to get MixMonitor to record in speex, but MixMonitor doesn't have an option for audio format. |
| 22:43.26 | Qwell | joesuffceren: no |
| 22:43.36 | joesuffceren | Qwell: thanks. I'll remove it |
| 22:43.43 | Tene | trying to record to .ogg results in vorbis, recording to .speex or .spx results in no recording. |
| 22:43.46 | Tene | Any ideas? |
| 22:44.08 | drmessano | KyleK: Most dont |
| 22:44.19 | Qwell | Tene: There is no speex format module, apparently |
| 22:44.38 | Tene | Qwell: "format module"? |
| 22:44.42 | trentcreek | Ass of now....$1 =347.000.00 |
| 22:44.43 | Qwell | What reads/writes files. |
| 22:44.51 | Tene | Ah. |
| 22:45.03 | Qwell | codec != format module |
| 22:45.06 | Tene | So, just not possible? OK. |
| 22:45.18 | Qwell | seems odd to me, but... |
| 22:45.40 | Qwell | (I would have bet money that we had one) |
| 22:51.05 | telnettech | need some odbc help with asterisk...anybody? |
| 22:51.17 | KyleK | can I start with an 'n' in extensions.conf like exten => xxx,n,Dial(SIP/xxxx)? |
| 22:52.13 | anthm | making a speex format is a challenge because the packets are variable length, there was once format module that has poor man's framing by writing the size of the packet in a 4 byte frame followed by that many bytes of audio but it was not exactly ideal |
| 22:53.09 | lesouvage | KyleK: No you have to start with a number (normally 1), the n counts from there. |
| 22:53.32 | telnettech | I am trying to connect to a remote mysql thru the ODBC connection for voicemail |
| 22:53.39 | anthm | the newest dev release of libsndfile has ogg so one format_sndfile would buy you dozens of formats |
| 22:55.23 | KyleK | telnettech: hows that working out? |
| 22:55.32 | telnettech | not well |
| 22:56.03 | telnettech | i have been comparing it with another site that has the same setup |
| 22:56.08 | *** join/#asterisk orpheee (n=u-applet@jem75-10-88-182-221-20.fbx.proxad.net) |
| 22:56.09 | telnettech | so i cant figure it out yet |
| 22:56.13 | orpheee | hello |
| 22:56.23 | KyleK | ah |
| 22:56.43 | KyleK | telnettech: i barely know what odbc is and does, otherwise i'd help |
| 22:56.59 | telnettech | thats fine.... |
| 22:57.31 | rhassing | telnettech, what did you configure so far? |
| 22:57.49 | rhassing | telnettech, maybe I can help |
| 22:57.59 | telnettech | http://pastebin.com/d4b54b9d1 |
| 22:58.09 | telnettech | that is my res_odbc.conf file |
| 22:58.36 | telnettech | http://pastebin.com/d3e1cccfd |
| 22:58.49 | telnettech | and that is my odbc.ini and odbcinst.ini files |
| 22:59.11 | KyleK | lesouvage: i guess i could start with a noop? |
| 22:59.28 | telnettech | http://pastebin.com/d29be2914 |
| 22:59.36 | telnettech | and this is the error that i get on CLI |
| 23:00.01 | telnettech | it was working before someone decided to throw a loop into the network sometime this morning |
| 23:00.20 | telnettech | I guess they were trying to test us agan today....this is par for this customer |
| 23:00.22 | lesouvage | KyleK: Yes something like exten => n,1,NoOp(Starting here) |
| 23:01.10 | lesouvage | good night all |
| 23:01.57 | KyleK | good night |
| 23:02.12 | orpheee | hello, i'm new on asterisk and i try to do an basic IVR: people press "*" and choose different choice : you can see my error there http://pastebin.com/d74cd2211 |
| 23:03.04 | M1s3ry | @orpheee, you have a space behind Goto. |
| 23:03.09 | M1s3ry | get rid of it. |
| 23:03.55 | jplank | anyone know off the top if their head what the polycom gmtoffset for BST is? I know its gmt+1, but I don't think it translates like that (considering US/Eastern is -18000) |
| 23:04.03 | M1s3ry | exten => *, 1, Goto (menu_choix, s, 1) should be exten => *, 1, Goto(menu_choix, s, 1) |
| 23:04.24 | rhassing | telnettech, Why do you put the username and password in the res_odbc.conf? |
| 23:04.42 | rhassing | telnettech, I think they should not be there |
| 23:04.57 | orpheee | ok M1s3ry i try |
| 23:05.41 | telnettech | that has always been there.......it worked before this morning |
| 23:05.43 | rhassing | telnettech, The server is reachable from the * ? |
| 23:05.52 | rhassing | on port 3306? |
| 23:05.59 | telnettech | when you say reachable what do you mean |
| 23:06.07 | telnettech | from the command prompt |
| 23:06.14 | telnettech | no |
| 23:06.38 | rhassing | telnettech, can you do a "telnet 192.168.15.2 3306" |
| 23:06.48 | telnettech | <PROTECTED> |
| 23:06.48 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 23:06.56 | rhassing | telnettech, you should get a response, or else it's a networking problem |
| 23:07.32 | *** join/#asterisk atraxxx (n=alex@65.183.213.66) |
| 23:07.43 | M1s3ry | Arg, I have an Asterisknow 1.5 install with a TE220p that's been handed to me by my boss. Somewhere along the lines of getting the te22op config in /etc/dahdi/system.conf, adding #include dahdi-channels.conf in chan_dahdi.conf, and then restarting asterisk/dahdi, chan_dahdi.conf no longer wants to load. I've tried all I can think of, and even installing dahdi from source and it still doesn't work. Anyone have an idea what to do? |
| 23:08.12 | telnettech | yes, I can telnet and I can ssh to the remote server |
| 23:08.40 | rhassing | telnettech, also on port 3306? |
| 23:08.48 | telnettech | I was able to get into the mysql thru the ssh and do the grant priviledge command |
| 23:09.06 | rhassing | M1s3ry, Does "dahdi show status" give any output? |
| 23:09.29 | orpheee | M1s3ry it'sok but now it tell me that http://pastebin.com/d534b87b1 |
| 23:09.40 | telnettech | i get a bad handshake connection thru port 3306 |
| 23:10.01 | rhassing | telnettech, but to make sure it is reachable from the * server you could do a telnet to port 3306 on the mysql server |
| 23:10.01 | M1s3ry | rhassing nay, dahdi does not exist in the cli commands, hence chan_dahdi.so does not load |
| 23:10.13 | orpheee | they are not all moment plz |
| 23:10.42 | telnettech | rhassing: I get mysql |
| 23:10.42 | telnettech | 5 |
| 23:10.42 | telnettech | 4.1.10asD8+md0`,sC2nWIFc?PS2Bad handshakeConnection closed by foreign host. |
| 23:11.04 | M1s3ry | orphee, looks like you have a space before the "s" try taking that out under menu_choi |
| 23:11.14 | orpheee | there :http://pastebin.com/d36aff3a2 |
| 23:12.14 | M1s3ry | orpheee same thing... you have a space in between "Playback" and "(<soundfile>)" |
| 23:12.18 | M1s3ry | try taking that out |
| 23:13.09 | M1s3ry | "Playback" is a command, whereas "Playback " (notice the space there) is not a command |
| 23:13.52 | rhassing | telnettech, Hmm, strange, I would say it should work... Maybe reload the module? To be honest I don't what else could be wrong... |
| 23:14.06 | M1s3ry | rhassing, I hate to harass you :p but any other ideas? |
| 23:14.52 | jplank | anyone in the UK? |
| 23:15.08 | M1s3ry | oh, or was your last statement directed at me? :) Ya I tried that first. unloading, loading, and reloadind the module doesn't do anything. |
| 23:16.23 | M1s3ry | lame, Module show like chan_dahdi.so shows that it's loaded, but dahdi is not a valid command in the CLI |
| 23:16.56 | rhassing | M1s3ry, no :) But maybe you could try to lod DAHDI first without to much config statements in it. |
| 23:17.43 | rhassing | M1s3ry, It should than at least load and give you the DAHDI_DUMMY |
| 23:19.20 | telnettech | thanks rhassing for your assistance anyways...gonna pass on to Tier IV....let them earn their pay |
| 23:19.53 | rhassing | telnettech, Sorry I couldn't help... |
| 23:20.03 | telnettech | np mate |
| 23:21.11 | M1s3ry | rhassing, are you talking about the card modules? the card is seen and the card module is installed. |
| 23:22.02 | rhassing | M1s3ry, no, just try using an empty chan_dahdi.conf file |
| 23:22.20 | jameswf | How do you release a product no one is apparently selling |
| 23:22.32 | rhassing | M1s3ry, That should load and give you just the dummy driver loaded |
| 23:23.01 | rhassing | M1s3ry, if it doesn't load, you might need to recompile... |
| 23:26.46 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 23:26.51 | M1s3ry | rhassing, ahhh... dahdi is back, so it was loaded as it stated.... now to get the configs in there |
| 23:27.16 | rhassing | M1s3ry, good job ;-) |
| 23:29.57 | *** join/#asterisk jake[work] (n=Jake@pool-173-52-144-183.nycmny.east.verizon.net) |
| 23:30.37 | jake[work] | will "noapic acpi=noirq" cause problems for my * install? i have a 2 t1 cards |
| 23:32.17 | *** part/#asterisk neal29 (n=np20433@nat/sun/x-c484fe5d35b22940) |
| 23:32.26 | orpheee | M1s3ry : ok for the space :) now.. my sounds http://pastebin.com/d1394ef7d |
| 23:33.08 | Qwell | orpheee: Type EXACTLY this. ls -l /var/lib/asterisk/sounds/hello-world.gsm.gsm |
| 23:33.25 | Qwell | Show me the output :) |
| 23:33.55 | KyleK | hahahaha |
| 23:35.34 | rhassing | orpheee, You should not give the extension at PlayBack. * will find the best extension available... |
| 23:35.48 | Qwell | rhassing: sure, take all the fun out of it.. |
| 23:35.55 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 23:35.59 | rhassing | Qwell, sorry ;) |
| 23:36.02 | Qwell | mutters something about feeding fish |
| 23:36.10 | orpheee | .gsm.gsm ? |
| 23:36.24 | KyleK | [May 4 09:16:17] WARNING[3367]: file.c:655 ast_openstream_full: File hello-world.gsm does not exist in any format <-- its looking for like hello-world.gsm.ul etc for some reason |
| 23:36.39 | KyleK | so just take the .gsm off |
| 23:36.56 | orpheee | -rw-r--r-- 1 root root 2508 jan 10 00:09 /var/lib/asterisk/sounds/hello-world.gsm |
| 23:37.10 | Qwell | orpheee: not what I said to type |
| 23:37.14 | KyleK | yea so Play(hello-world) |
| 23:38.20 | rhassing | Qwell, Let him listen to Allison as well :) |
| 23:39.25 | orpheee | hey hey good rhassing i left my extension and that good thks i continue my test ^^ |
| 23:46.13 | ruben23 | hi |
| 23:46.51 | ruben23 | i have a public asterisk server...how do i register my SIP client softphones on remote |
| 23:49.14 | ruben23 | its not behin NAT. |
| 23:54.48 | *** join/#asterisk lizor (n=liz@rigel.midgard.org) |
| 23:54.57 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 23:58.37 | KyleK | ruben23: so you're registering a softphone somewhere with the asterisk server? |